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Author Topic: Resample to 44.1 or record at 44.1 (midside Q)  (Read 13372 times)

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Offline Lil Kim Jong-Il

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Re: Resample to 44.1 or record at 44.1 (midside Q)
« Reply #15 on: December 28, 2008, 02:08:42 AM »
Prove? I have (as yet) never seen any properly conducted experiment that showed people can tell the difference between any sample rate above 44.1 kHz or bit depth above 16 (all else being equal), the data points to the contrary. Got links?

digifish

I find it amusing that the referenced article is offered in two formats "(PDF-615KB)" and "(HI-RES PDF-9.4MB)". 

Unfortunately the article is available to AES members only.

In my playback system I have done AB testing and while sample rate doesn't seem to make a difference, 24-bit vs 16-bit does.  Could be the authoring of the 24-bit material or maybe the DAC does a better job with 24-bit sources.  I don't know why the 24-bit sounds different to me but I've always allowed that maybe I imagine it.  I've read several reviews of up-sampling DACs that stated the up-sampled 16-bit sources sound every bit as good as the native 24-bit material.  In anycase, I use tubes so the noise level and frequency range of my gear is above and below the respective thresholds for 16-bit material anyway. 
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Offline DSatz

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Re: Resample to 44.1 or record at 44.1 (midside Q)
« Reply #16 on: December 28, 2008, 01:11:15 PM »
digifish_music, bit depth (e.g. 16- vs. 24-bit) is a clear-cut, practical matter about which there is no great controversy or mystery. It corresponds directly to the dynamic range of a recording.

If you don't know in advance what the maximum sound pressure level at a performance will be, and you're there to record it, 24-bit recording offers a definite advantage since it allows you to set your levels very conservatively. Then once the concert is over and you know what the peak levels actually were, you can renormalize the recording and dither down to 16 bits. If the maximum levels were, say, -8 dBFS then you can have a nearly 8 dB improvement in the dynamic range of the finished product, as compared with recording the concert at 16-bit resolution using the same level settings.

I record classical concerts professionally, and am generally the only audio engineer at most of the events I record. To me this approach seems far more responsible (though less exciting, perhaps) than the old way of trying to guess and occasionally, guessing wrong which I did for 35+ years before I changed over to 24-bit recording for most events.

In my opinion 16 bits is usually more than enough for domestic audio, but occasionally some program material comes along (e.g. percussion ensembles) which can really exploit almost its full range, so I'm glad to have it. And for me, 24-bit recording is the low-stress way to get an optimal 16-bit recording--it's not an end in itself. As far as a concrete proof (or demonstration) of this is concerned, that is actually not difficult at all, and if you'd like to arrange something I'd be glad to help. But it's simply not correct to claim that there has been no proof of it up to now--it is an elementary principle which has been very well known for decades.

The question of sampling rate (sampling frequency) is rather a different sort of topic to discuss critically, and I don't want to try to go there in the same message. I'd rather try to establish some clarity about bit depth, since that really is pretty simple.

The only strong caveat with bit depth is to remember that there is always noise in any audio system, and the actual signal-to-noise ratio of any real-world recording (i.e. a rendering of acoustic events, rather than something algorithmically generated) will always be distinctly less than the maximum which the 24-bit format would allow. 24 bits = almost 143 dB dynamic range, which exceeds by far the dynamic range of the quietest physically possible microphones, mike preamps and A/D converters (let alone the quietest ones which are actually available). Thus by definition any 24-bit recording has less than 24-bit "actual" resolution (i.e. signal-to-noise ratio), and it's important not to confuse the container with its contents.

--best regards
« Last Edit: December 28, 2008, 01:16:54 PM by DSatz »
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Offline boojum

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Re: Resample to 44.1 or record at 44.1 (midside Q)
« Reply #17 on: December 28, 2008, 01:40:27 PM »
Nor am I convinced.  I have been down this road many, many times.  Oh, yes.  Green magic markers on the edges of CD and all the rest of the hoo-ha touted by folks "who know" or claim they know or print expensive and revered magazines.  Remember the Shin-Mook wooden dots?  Made everything sound better.  Oh, yeah, right.

There are a couple of other boards, pro boards, which are very strong for recording at 96kHz or higher.  Indeed, I have seen 48 kHz pooh-poohed on this board as "not good enough for serious music recording."  But I, too, have never seen anyone put forward double blind ABX tests to support their positions.  They just "know" they can hear the difference without testing.  Well, good for them.  I am still not sure.

Cheers
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Offline digifish_music

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Re: Resample to 44.1 or record at 44.1 (midside Q)
« Reply #18 on: December 28, 2008, 05:21:05 PM »
digifish_music, bit depth (e.g. 16- vs. 24-bit) is a clear-cut, practical matter about which there is no great controversy or mystery. It corresponds directly to the dynamic range of a recording.

I agree, but I also wasn't talking about production/post production, I was talking about listening to the final mix. Can you find any scientific paper that shows people can reliably tell a 24 bit from 16 bit derivation of the same 24 bit recording under controlled (normal level) listening conditions?

Evidence against.

Quote from: DSatz
24 bits = almost 143 dB dynamic range, exceeds by far the dynamic range of the quietest physically possible microphones, mike preamps and A/D converters (let alone the quietest ones which are actually available). Thus by definition any 24-bit recording has less than 24-bit "actual" resolution (i.e. signal-to-noise ratio), and it's important not to confuse the container with its contents.

Indeed again, in reality the bit depth of most "24 bit" D/A's is 18-20 bits ~20 is approx the limits imposed by electronic components at room temperature, and as good as it is ever going to get. IMO there is still much more sonic milage to be gained by focusing on making the analog stages better (so that a good D/A or A/D can make use of the bits it has).

digifish
« Last Edit: December 28, 2008, 09:14:03 PM by digifish_music »
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Offline digifish_music

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Re: Resample to 44.1 or record at 44.1 (midside Q)
« Reply #19 on: December 28, 2008, 05:33:25 PM »
They just "know" they can hear the difference without testing.  Well, good for them.  I am still not sure.

Cheers

Nor should you be. Despite years of research there has been very scant evidence that people respond to frequencies above 20 kHz (much less if you are older than 20) and many studies that show there is no effect. The recent 'The Emperor's New Sampling Rate, summarises one of the better (recent) studies.

Also it's worth having a read of this...

Perceptual Discrimination between Musical Sounds with and without Very High Frequency Components, NHK Labs.

and this

David Griesinger's (of Lexicon) Experiments in Ultrasonics

IMO it's time to put this to bed and move on. After all the reason for the push has been equipment manufacturers trying to stop their market stagnating and heading for low-cost where there are small margins. Record labels trying to replicate the 80's boom where people re-purchased all their old LPs on CD, they were hoping for it to happen again with DVD audio or SHCD, fat chance.

I think one of the big problems for the average punter is that they buy some new gear that has 24 bit @ 96 kHz sampling and it does sound better than their old gear, so they assume it's all about the bit depth. In reality it's got more to do with the better analog design and possibly better quality D/A or A/D, but not the # bits or kHz.

digifish
« Last Edit: December 28, 2008, 07:29:24 PM by digifish_music »
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Offline DSatz

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Re: Resample to 44.1 or record at 44.1 (midside Q)
« Reply #20 on: December 29, 2008, 12:21:22 AM »
digifish_music asked:

> Can you find any scientific paper that shows people can reliably tell a 24 bit from 16 bit derivation of the same 24 bit recording under controlled (normal level) listening conditions?

If I were to put forth a fancy-sounding claim that a person with normal hearing can perceive a 1 kHz tone at 65 dB SPL, it would be rather silly--not because it's wrong, but because everybody already knows it. If I were to write up my claim as a research paper and describe clinical trials with controlled, double-blind listening experiments, I'd have some difficulty getting it published. It's the old "dog bites man" vs. "man bites dog"--it wouldn't have any news value.

Similarly, if someone were to carry out controlled listening tests on telling a 24-bit system apart from a 16-bit system, I can see very little reason for anyone to publish the result. The outcome is easily predictable from basic knowledge that's been available for decades, and it would just be a waste of journal space.

In a very quiet playback environment (< 20 dB ambient SPL for example), if you set the gain of a playback system so that the peak levels are at 120 dB SPL, then the noise floor of a dithered 16-bit system will be discernably higher than the ambient noise level. (I chose 120 dB SPL because it is close to the highest SPL that I have directly measured with a number of the opera singers I have recorded; let's say I want to play back their singing at precisely realistic levels.) At an identical playback gain, however, the noise floor of a dithered 20-bit system would fall distinctly below the ambient noise level. And so would that of a dithered 24-bit system. This difference in noise floor would be obliterated in the presence of any program material, but would be discernable in the "silence" between tracks.

You want to doubt it? Doubt it--but try it some time, please. The noise floor of a dithered 16-bit system is extremely low when compared to analog tape of all kinds, but it isn't so low that it is never, ever audible. With 20- and 24-bit systems the noise floor is distinctly lower, and there's simply no reason for that difference to be inaudible to a consumer when the playback gain is set high enough.

--best regards
« Last Edit: December 29, 2008, 05:05:25 PM by DSatz »
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Offline digifish_music

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Re: Resample to 44.1 or record at 44.1 (midside Q)
« Reply #21 on: December 29, 2008, 01:41:30 AM »

In a very quiet playback environment (< 20 dB ambient SPL for example), if you set the gain of a playback system so that the peak levels are at 120 dB SPL, then the noise floor of a dithered 16-bit system will be discernably higher than the ambient noise level.


I don't doubt it (indeed the paper I linked too says the same, "at very elevated levels"), but then those are not 'normal' listening conditions (so that peaks are 120 dB). It's very hard to find spaces with 20 dB background noise and very few recordings are made that use 120 dB dynamic range...

The paper I pointed to suggests the opposite about the differences between 44.1@16 bit vs others -

http://www.aes.org/e-lib/browse.cfm?elib=14195

24 & 16 bit are indistinguishable over the range of 'normal' & 'loud' listening levels.

For production then 24 bit makes a lot of sense, for playback not at all.

So someone did 'waste space' - Abstract:

"Claims both published and anecdotal are regularly made for audibly superior sound quality for two-channel audio encoded with longer word lengths and/or at higher sampling rates than the 16-bit/44.1-kHz CD standard. The authors report on a series of double-blind tests comparing the analog output of high-resolution players playing high-resolution recordings with the same signal passed through a 16-bit/44.1-kHz “bottleneck.” The tests were conducted for over a year using different systems and a variety of subjects. The systems included expensive professional monitors and one high-end system with electrostatic loudspeakers and expensive components and cables. The subjects included professional recording engineers, students in a university recording program, and dedicated audiophiles. The test results show that the CD-quality A/D/A loop was undetectable at normal-to-loud listening levels, by any of the subjects, on any of the playback systems. The noise of the CD-quality loop was audible only at very elevated levels."

digifish 
« Last Edit: December 29, 2008, 05:47:17 AM by digifish_music »
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Offline DSatz

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Re: Resample to 44.1 or record at 44.1 (midside Q)
« Reply #22 on: December 30, 2008, 09:00:20 AM »
digifish, in that study (written by two guys I know from Boston, and definitely a fun read), recordings made at sampling rates higher than 44.1 kHz were converted down to 16-bit/44.1 kHz. The study didn't show that no sound quality was lost--only that the subjects in that study, under particular listening conditions, didn't reliably perceive any loss in sound quality.

That's a nice data point, but it doesn't paint a complete, permanent picture. Sample populations are always finite, people can learn to hear some sonic defects that they couldn't hear before, and another well-constructed, controlled study could perhaps be published some day in which one or more people do hear a difference to a reasonable degree of statistical confidence. I don't personally expect that--but that's as much a matter of personal belief as it is of science.

In any case some people here (including me) like to go beyond merely "loud" to VERY LOUD at times. The study didn't deal with that, and it has a direct bearing on the issue of bit depth.

If someone feels that a particular 96 kHz recording sounds better to him (whatever "better" may mean to him at the time) than recordings from whatever 44.1 kHz systems he happens to have heard before, I feel that I have to leave that alone. I may have big doubts, but I simply don't know of any way to tell a person that for reason x, y and/or z he can't possibly be right about his own opinion, without insulting him and making a fool of myself. Maybe he hears certain things better than I do--I've definitely met some people who do.

--best regards
« Last Edit: December 30, 2008, 12:45:47 PM by DSatz »
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Offline SmokinJoe

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Re: Resample to 44.1 or record at 44.1 (midside Q)
« Reply #23 on: January 06, 2009, 09:30:51 PM »
I still find it interesting... we go to a rock show.  Guitarist plays guitar into amp.  Amp is mic'ed with SM57 with 'frequency response' or 40-15,000 hz.  Vocalist sings into SM58 with similar response.  All this goes into mixing board, into stacks, we sit 100' from stacks with "good mics" with response from 20-20,000hz to record the room.  A 44.1k bitrate really will capture all of the above.  Maybe the lady shrieking next to my Earthworks (up to 30,000 hz) will benefit from a higher sample rate recording, but she's not that important to me.  Still, the taper next to me will insist on recording at 96k, even though there is nothing on that wav file higher than 20khz.

Recording a symphony (or instrumental bands stage lip??) without all the above intermediate products might actually get some value  from higher bitrates, but it's pure overkill unless your mics can pick it up, and most of our mics drop off quickly after 20khz.

I agree that recording and archiving at 24bit are well worth it.  Someday we may look back on those raw tapes like the high quality SBD reels from early Grateful Dead and be glad we kept them.
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Offline boojum

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Re: Resample to 44.1 or record at 44.1 (midside Q)
« Reply #24 on: January 06, 2009, 11:43:45 PM »
I still find it interesting... we go to a rock show.  Guitarist plays guitar into amp.  Amp is mic'ed with SM57 with 'frequency response' or 40-15,000 hz.  Vocalist sings into SM58 with similar response.  All this goes into mixing board, into stacks, we sit 100' from stacks with "good mics" with response from 20-20,000hz to record the room.  A 44.1k bitrate really will capture all of the above.  Maybe the lady shrieking next to my Earthworks (up to 30,000 hz) will benefit from a higher sample rate recording, but she's not that important to me.  Still, the taper next to me will insist on recording at 96k, even though there is nothing on that wav file higher than 20khz.

                     <snip>


My take exactly.  Just how good does your gear have to be to record 57's and 58's??  Acoustic recordings are a whole other ball game, but recording 57's and 58's out of a pair, or more, or stack, well, there is a lot of gear which would do the job pretty well.  And not all of it is expensive.     8)
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Offline prof_peabody

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Re: Resample to 44.1 or record at 44.1 (midside Q)
« Reply #25 on: January 23, 2009, 05:43:55 PM »
Question -

You record a 4 kHz signal at a sampling rate of 12 kHz.  Each single cycle will only have 3 points representing the waveform of one cycle and you reconstruct it using some form of interpolation.  If you increase the sample rate to 24 kHz, you will have 6 points describing the waveform of one cycle, which is used to interpolate.  Which one will give a better representation of the initial signal?

Offline boojum

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Re: Resample to 44.1 or record at 44.1 (midside Q)
« Reply #26 on: January 23, 2009, 06:49:12 PM »
If what you are saying is valid it would seem the higher the frequency the fewer sampling points and the obviously degraded signal.  Low notes would be accurate, high notes would not.  A 22.05 kHz note would have two "describing points", a 10 Hz note would have 4.41 thousand.  Does that sound right???
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Offline prof_peabody

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Re: Resample to 44.1 or record at 44.1 (midside Q)
« Reply #27 on: January 23, 2009, 08:02:53 PM »
If what you are saying is valid it would seem the higher the frequency the fewer sampling points and the obviously degraded signal.  Low notes would be accurate, high notes would not.  A 22.05 kHz note would have two "describing points", a 10 Hz note would have 4.41 thousand.  Does that sound right???

Yeah. 

Ok, so next question.  The human ear can hear 20 Hz to 20 kHz, for the sake of this, let's say worthwhile sound tops out at 16 kHz.  If you sample at 44.1 kHz, the frequencies near 16 kHz are only going to have ~3 sampled points to represent the waveform right?  But at the same time, lower frequency sounds are going to have >3 points, so they are probably really well sampled.  So does this mean recording at 48 kHz might improve the representation of the waveform for higher frequency sounds?

Offline Will_S

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Re: Resample to 44.1 or record at 44.1 (midside Q)
« Reply #28 on: January 23, 2009, 08:15:05 PM »
Question -

You record a 4 kHz signal at a sampling rate of 12 kHz.  Each single cycle will only have 3 points representing the waveform of one cycle and you reconstruct it using some form of interpolation.  If you increase the sample rate to 24 kHz, you will have 6 points describing the waveform of one cycle, which is used to interpolate.  Which one will give a better representation of the initial signal?

Unless something is wrong with your interpolation scheme, both will recreate the 4 KHz signal perfectly (well, perfectly subject to bit-depth constraints).

Edit:  Imperfect analogy:  Does having 3 points define a line any better than 2 points?
« Last Edit: January 23, 2009, 08:57:12 PM by Will_S »

Offline prof_peabody

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Re: Resample to 44.1 or record at 44.1 (midside Q)
« Reply #29 on: January 24, 2009, 01:16:36 AM »
Question -

You record a 4 kHz signal at a sampling rate of 12 kHz.  Each single cycle will only have 3 points representing the waveform of one cycle and you reconstruct it using some form of interpolation.  If you increase the sample rate to 24 kHz, you will have 6 points describing the waveform of one cycle, which is used to interpolate.  Which one will give a better representation of the initial signal?

Unless something is wrong with your interpolation scheme, both will recreate the 4 KHz signal perfectly (well, perfectly subject to bit-depth constraints).

Edit:  Imperfect analogy:  Does having 3 points define a line any better than 2 points?

The answer to your analogy is yes, 3.

My understanding is that there are a few different interpolation functions in a DAC.

http://en.wikipedia.org/wiki/Digital-to-Analog_Converter#DAC_types

So which one is the best?  Which one will give you the best interpolation of those three points that deffine your wave?

 

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