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Real-world comparisons of 24-bit and 16-bit music from Linn Records -
JimC
post Aug 17 2009, 15:45
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Hi Folks,

For anyone who is interested in doing a direct real-world comparison between 24-bit, 16-bit and MP3, we are starting a little 24-bit download series via Twitter.

First up is some awesome Jazz from Alyn Cosker.

Download the full 16bit and 24bit FLAC versions of this track for free and have a listen. I think you will be pleasantly surprised! Click the little down arrow on the right hand side to download each track, and make sure you preview them through a DAC or soundcard capable of reproducing 24-bit/96Khz.

Get the tracks here

Follow us on Twitter too if you would like to keep up with it and hear some more. Coming up in the series we have some classical recordings (at 24/192kHz!) some great rock and pop from Maeve O'Boyle, and as yet unreleased new work from jazz great Claire Martin!

Enjoy!


Jim - Linn Records
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pdq
post Aug 17 2009, 16:49
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This is not a copyright violation is it?
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rpp3po
post Aug 17 2009, 16:51
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I guess not if the a guy from the label is posting them.

It would be better if the tracks were downloadable, though (or am I just not getting how to do this?). Else you never know wether an audible difference is just due to improper quantization/dithering. I would like to be able to do that myself, before coming to false conclusions.

Can you make that possible, Jim?

Edit: Downloading is possible.

This post has been edited by rpp3po: Aug 17 2009, 16:57
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tpijag
post Aug 17 2009, 16:58
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Down arrow on the right side will download.

terry
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JimC
post Aug 17 2009, 17:09
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Hi Folks,

Yeah the small arrow on the right is a little subtle! Playing them online will give you the chance to contrast the downloaded files to a 128k MP3 stream - which is the what a lot of people listen to regularly, and in some cases exclusively! Once you have heard the beauty of the 24-bit Studio Master files its hard to go back to anything else.

Some really impressive recordings to come... so keep an eye out.

(and of course we are the copyright owners, so it's all completely guilt free!)

Jim
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rpp3po
post Aug 17 2009, 17:15
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Thanks for making this available! I'll happily give it a try.

It's pretty courageous to make this public. Many people might not be able to hear a difference in a proper ABX setup. But if at least some do, you are going to win the whole pot. wink.gif
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Axon
post Aug 17 2009, 18:22
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Take a very long look at the waveform nulls before trying to ABX anything - IIRC, I was unable to null the samples to under perhaps -30dBFS or so. Some waveforms are visually different well above quantization noise. I'm not sure yet if this is some meaningless artifact of the conversion or if it represents a substantial DSP difference, but the sheer magnitude of the waveform differences does not leave me with a swell feeling.

EDIT: See below.


This post has been edited by Axon: Aug 17 2009, 21:53
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rpp3po
post Aug 17 2009, 19:07
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Let me first say this: This 24/96 track from Linn is an excellent recording, certainly well worth a buy!

But what most folks at Hydrogenaudio.org are interested in is mainly the question wether the 24/96 storage format is able to make an audible difference vs. Redbook, when all rules of proper mastering have been followed.

As the samples have been provided by Linn, objective comparison is not possible, mainly due to clipping (and maybe also because of aspects Axon has touched). When you set your output sample rate to 96kHz Linn's Redbook sample cannot be SRC'ed at high quality without clipping. This will produce audible differences that are not caused by limitations of the 16/44.1 storage format.

In the following I will describe a procedure, that you can employ to produce your own highest quality Redbook masters from Linn's HiRez sample and prepare them for ABX testing.

1. Convert the Flacs to WAV.
2. Download the excellent Sox from sox.sourceforge.net and place the binary in the same folder as the Bheki WAVs.

3. Generate your own Redbook master from Linn's HiRez WAV (VHQ SRC with intermediate phase low-pass, noise shaped dither, 1db gain reduction to prevent clipping):

CODE
./sox Bheki.wav -b 16 Bheki-16bit-custom.wav gain -1 rate -v -I 44100 dither -s


4. Take the down-converted Redbook sample and create an upsampled 96kHz version from it. This eliminates any side effects for the later ABX comparison without increasing the actual resolution of the content. Again VHQ intermediate phase SRC:

Edit: Fixed wrong copy & paste:
CODE
./sox Bheki-16bit-custom.wav -b 24 Bheki-16bit-custom-abx.wav rate -v -I 96000


5. Apply the same -1db gain to the HiRez sample, that you have applied to the Redbook sample to prevent clipping, without touching anything else:

CODE
./sox Bheki.wav -b 24 Bheki-abx.wav gain -1


Now ABX the HiRez Bheki-abx.wav vs. the LowRez Bheki-16-bit-custom-abx.wav and check if you can hear a difference.

If you're interested in how Linn's 16-bit compares to the HiRez version without the effect of clipping, try the following:

CODE
./sox Bheki-16bit.wav -b 24 Bheki-16bit-abx.wav gain -1 rate -v -I 96000


Then ABX the HiRez Bheki-abx.wav. vs. Linn's LowRez Bheki-16-bit-abx.wav.

As a last step ABX Linn's provided 16 bit mixdown vs. their HiRez sample.

I would provide these samples myself but that obviously would be copyright violation.

This post has been edited by rpp3po: Aug 17 2009, 23:00
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krabapple
post Aug 17 2009, 19:20
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Here's another publicly-available pair of clips , offered by the engineer for comparison of 16/44 to 24/96

http://www.soundkeeperrecordings.com/format.htm
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Kitsuned
post Aug 17 2009, 20:03
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Maybe my setup, environment, ears (or a combination of all) aren't all that good because I hear no discerning differences by any of the files offered for comparision in this topic. I didn't even bother doing a blind test...I heard no differences knowing which track was which and would fail an abx easily.

Am I missing something or is there no real difference in the files? Perhaps someone could help me, or I'm one of the believers that 16-bit, 44khz is good enough for any equipment I'll ever own. smile.gif


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rpp3po
post Aug 17 2009, 20:19
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BTW, Linn's samples are of different length and not time-aligned. One should correct that before comparing with an instant-switching capable setup (as Foobar's ABX component). I'll start my round of testing now.

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pdq
post Aug 17 2009, 20:25
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@Kitsuned: Don't worry. I expect few if any will be able to distinguish these in a proper ABX test.
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rpp3po
post Aug 17 2009, 21:12
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Man, that's hard. I could swear that the low resolution sample sounded muffled compared to the high resolution version, when playing them separately in iTunes! laugh.gif No joke. But loaded into the ABX comparator the differences seem to be gone. I'm at 50/50 after 4 tries and now my ears ring due to the relatively high volume.

I'll continue as soon as the fatigue is gone.
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2Bdecided
post Aug 17 2009, 21:32
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I love Linn records, and Claire Martin especially.

Thanks for the free music!

Cheers,
David.
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Axon
post Aug 17 2009, 21:46
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False alarm on my front - the nulling problems I was describing seem to have gone away entirely after upsampling to 384khz, attenuating -6db, sample-aligning, then downsampling back to 44.1khz. Comparing Bheki-16bit to Bheki after all that results in a frequency response variation of only +- 0.003db from 0-15khz and only +- 0.015db above that.

Note, however, that the 16-bit version appears to be attenuated by 0.165db relative to the high res version. This is just barely on the cusp of what I might consider audible.
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rpp3po
post Aug 17 2009, 21:49
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I give it up, over 80% probability that I'm guessing after 10 trials. And I really did give it a chance: loud volume, several ranges tested, long runs. And the equipment wasn't bad either, a Benchmark DAC1 and a pair of Westone UM2, which tend to be my most revealing headphones.

I just compared the high resolution track against my own 16 bit version, since Linn's is not aligned and there's not much (probably nothing) you can do better than the above procedure. So after all that I don't buy the 24/96 advantage!

Still this is a nicely done record. They have used some kind of tube effect, that adds nice distortion to the louder amplitudes as some piano highlights, which makes it very pleasant to hear. Stereo image and setup are also well done. It didn't have to be mastered that loud considering it is a 24 bit record, but where else do you get that nowadays?

This post has been edited by rpp3po: Aug 17 2009, 23:13
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evereux
post Aug 17 2009, 22:28
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I tried ABXing the two files and failed. (MAudio Audiophile 24/96 > Arcam A80 > HD600)

Very nice sounding music though, thanks JimC. Something for me to investigate further.


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AtaqueEG
post Aug 18 2009, 06:03
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QUOTE (gmwiz05 @ Aug 17 2009, 17:22) *
For some reason the 24-bit one sounds like it has a more in-depth stereo feel to it than the 16-bit one. Now this is only noticed when listening at the original sound level, the bass seems to have more clarity too. blink.gif



Maybe it is because you are listening with full knowledge of which is which?

I am sure ABX will make those differences go away.


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MLXXX
post Aug 18 2009, 16:01
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In another HA thread, it has been found that for a given sample rate, dithering a 24-bit original with appropriate noise shaped dither to 16 bits makes no audible difference, for music recorded and played back at normal listening levels. (See http://www.hydrogenaudio.org/forums/index....st&p=626692 )

It is very hard to make comparisons of 96KHz and 48KHz as a soundcard may operate slightly differently when called upon to process a different sample rate. Such differences as may exist for a human ear, seem likely to be extremely subtle. As a workaround, I note rpp3po's suggestion:
QUOTE (rpp3po @ Aug 18 2009, 04:07) *
4. Take the down-converted Redbook sample and create an upsampled 96kHz version from it. This eliminates any side effects for the later ABX comparison without increasing the actual resolution of the content.

Tonight I had a quick informal listen and I found the 24-bit version did sound slightly better; but then I realised the 24-bit version sounded slightly louder. As Axon has mentioned, 0.165db will be barely on the cusp of what could be considered audible:
QUOTE (Axon @ Aug 18 2009, 06:46) *
Note, however, that the 16-bit version appears to be attenuated by 0.165db relative to the high res version. This is just barely on the cusp of what I might consider audible.
Some people could detect it consciously. Others would not be able to detect it at all.

It is well known empirical result that a sound file that is slightly louder will seem slightly "better", all other things being equal.

QUOTE (rpp3po @ Aug 18 2009, 06:49) *
I give it up, over 80% probability that I'm guessing after 10 trials. And I really did give it a chance: loud volume, several ranges tested, long runs. And the equipment wasn't bad either, a Benchmark DAC1 and a pair of Westone UM2, which tend to be my most revealing headphones.

I just compared the high resolution track against my own 16 bit version, since Linn's is not aligned and there's not much (probably nothing) you can do better than the above procedure. So after all that I don't buy the 24/96 advantage!

Still this is a nicely done record. They have used some kind of tube effect, that adds nice distortion to the louder amplitudes as some piano highlights, which makes it very pleasant to hear. Stereo image and setup are also well done. It didn't have to be mastered that loud considering it is a 24 bit record, but where else do you get that nowadays?
I have not gone to the trouble of preparing files in the manner rpp3po has described, and listening to them intently. Has anyone else gone through such an exercise? If so, what was the ABX result?

This post has been edited by MLXXX: Aug 18 2009, 16:12
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rpp3po
post Aug 19 2009, 01:48
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Common guys, I have just one pair of ears. Anyone else? So many people swear by 24/96. Convince me that it isn't just marketing hype.
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Cavaille
post Aug 19 2009, 09:38
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I did an ABX but I did it differently than before. I lowered the 24/96 for 0.07 dB in volume. That way it matched the volume of the 16/44.1 track. The latter track was a wee bit longer, compared to that the 24/96 track lacked a few samples at the beginning of the track - needless to say, that I added them by copy/paste until the tracks were perfectly aligned (with sample accuracy). I didnīt do any upsampling - that would have changed the results. My external interface very neatly switches the samplerates around without me knowing it (no clicks involved) and foobar adds a pause when every track (A, B, Y or X) is changed.

This is the result:

CODE
foo_abx 1.3.4 report
foobar2000 v0.9.6.7
2009/08/19 10:14:17

File A: C:\Users\Eunice\Desktop\Bheki-16bit.wav
File B: C:\Users\Eunice\Desktop\Bheki.wav

10:14:17 : Test started.
10:15:46 : 01/01  50.0%
10:16:38 : 01/02  75.0%
10:18:57 : 02/03  50.0%
10:19:57 : 03/04  31.3%
10:21:22 : 04/05  18.8%
10:21:41 : 05/06  10.9%
10:21:58 : 05/07  22.7%
10:23:48 : 06/08  14.5%
10:24:46 : 07/09  9.0%
10:26:30 : 08/10  5.5%
10:27:57 : 09/11  3.3%
10:28:10 : Test finished.

----------
Total: 9/11 (3.3%)


I actually had a hard time figuring out where the differences were. I finally settled with listening to transients, in particular, the percussion set which is more transparent with the 24/96 file. The 16/44.1 track seems to be a bit more compact & flat in the soundstage and offers less "bite" and "air" but the differences are very subtle - as they are always with 24/96 material.

EDIT: I forgot the equipment I used. E-MU 0202 USB with a Sennheiser HD-600 directly fed from it. foobar uses ASIO.

This post has been edited by Cavaille: Aug 19 2009, 09:42


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rpp3po
post Aug 19 2009, 10:03
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QUOTE (Cavaille @ Aug 19 2009, 10:38) *
I didnīt do any upsampling - that would have changed the results. My external interface very neatly switches the samplerates around without me knowing it (no clicks involved) and foobar adds a pause when every track (A, B, Y or X) is changed.


As far as I know the E-MU 0202 USB does not switch sample rates automatically, but employs Windows' mediocre quality realtime-SRC to convert to the rate set in the E-MU USB Audio Control Panel. So what you are testing is not a purer comparison, but HiRez vs. Windows' simplistic realtime-SRC. That alone would already be a comprehensible source of difference, but on top of that you get the problem I have described above. The 16 bit track, as it comes from Linn, cannot be converted to a higher sample rate (wether realtime or HQ) without resulting in several hundred clipped samples, so it must be attenuated prior to conversion.

Attenuation (of a 16 bit track) in addition requires re-dithering, which again raises the noise floor, so the approach to generate your own high quality master from the 24/96 source, with the best possible SRC and (only once applied) shaped dither, is the cleanest path to follow.

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2Bdecided
post Aug 19 2009, 10:12
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QUOTE (rpp3po @ Aug 19 2009, 01:48) *
Common guys, I have just one pair of ears. Anyone else? So many people swear by 24/96. Convince me that it isn't just marketing hype.
Starting with the 24/96 file, lossyWAV keeps 9.2 bits on average (default settings - generally transparent).

I looked at a specific section, which did not include any silence, and the most it kept was 13 bits, while the least it kept was 7 bits.


Compared with the Linn releases I have from a decade or more ago, that's a very "loud" recording with a lot of DRC and a lot of samples near digital full scale. I know it's not smashed to pieces 21st century pop, but it's not a pure / audiophile / minimalist recording letting you hear the true dynamics of the music or instruments.


If you're going to ABX 24/96 vs 16/44, I think you've got to generate both yourself, like this.

1. use the provided 24/96 as a source (call that A)
2. halve the volume (23 bits! wink.gif ) (call that B)
3. resample (B) to 16/44 (call that C) using fb2k or whatever
4. ABX (B) vs Đ
5. resample Đ back to 24/96 (call that D)
6. ABX (B) vs (D)

None of these are the provided files - but the provided files are unsuitable for ABX because of the level difference, and the possibility of clipping in any standard oversampled DAC. (In my simulation, it's inevitable that there will be clipping at least once, unless there's headroom above 0dB FS in the oversampling process).

If you can pass 6, then there's a real difference when replayed in your system. If you can pass 4 but not 6, then there's a difference in the performance of your sound card / DAC, but it's only due to the limitations of your DAC, not fundamental to the difference between 44/16 and 24/96.

Cheers,
David.

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Cavaille
post Aug 19 2009, 10:16
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QUOTE (rpp3po @ Aug 19 2009, 11:03) *
As far as I know the E-MU 0202 USB does not switch sample rates automatically, but employs Windows' mediocre quality realtime-SRC to convert to the rate set in the E-MU USB Audio Control Panel.
Oh, but it does. Completely automatic. One of the few USB interfaces that can do so (the 0404 USB does the same). Remember, the 0202 is one of the few asynchronous interfaces out there with complete control over the USB connection. Just look at the manual, you can download it for free from E-MU. It also offers a completely useless tool called "E-MU USB Audio Application" where it shows the actual samplerate and USB connection used. I did not use it in this test since I would have been able to tell which track was which without even listening. I never use it all - saving resources. wink.gif

QUOTE (rpp3po @ Aug 19 2009, 11:03) *
The 16 bit track, as it comes from Linn, cannot be converted to a higher sample rate (wether realtime or HQ) without resulting in several hundred clipped samples, so it must be attenuated prior to conversion. Attenuation again requires re-dithering, which again raises the noise floor, so the approach to generate your own high quality master from the 24/96 source is the cleanest path to follow.
Exactly. Thatīs why I used the 24/96 track for editing and left the 16/44.1 completely unspoiled. The quantization noise resulting from editing the 24/96 file should be at around -121 dB if Iīm not mistaken - and I did not use dithering when saving the edited and volume attentuated 24/96 track.


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rpp3po
post Aug 19 2009, 10:18
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QUOTE (2Bdecided @ Aug 19 2009, 11:12) *
1. use the provided 24/96 as a source (call that A)
2. halve the volume (23 bits! wink.gif ) (call that B)
3. resample (B) to 16/44 (call that C) using fb2k or whatever
4. ABX (B) vs Đ
5. resample Đ back to 24/96 (call that D)
6. ABX (B) vs (D)


What's the point of repeating above procedure with the difference of halving the volume (-1db would be sufficient) and omitting dither? Both aspects are additional source of error.
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