Old 08-11-2010, 09:36 AM   #1
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Default Audio Myths and DAW wars :) part 1

Here's a little something for you all

Audio Myths & DAW Wars

Three things you need to know about audio quality
Research has shown that for music distribution, 16 Bit @ 44.1 kHz (CD standard) is indistinguishable from 24 Bit @ 196 kHz in a sample of over 550 listeners. In other words, more Bits and higher Bit-rates are not going to improve the quality of your tracks.

There are many traps for young players when comparing audio from two DAWs, make sure you know what they are, discussed below.
The world is full of marketing departments trying to convince you that equipment and specifications can substitute for talent & hard work. This is not true, the 'performance' transcends the medium every time. The performance includes musicianship, vocals, orchestration, arrangement and the mixing decisions. These are all under your control and have little to do with the DAW software you use or plugins you have.

Audio quality, the eternal quest

Spend time on any forum devoted to any Digital Audio Workstation (DAWs) or music production and you will see users making unsubstantiated claims about the audio quality of this or that DAW. Protagonists will say a given DAW is clearly and audibly superior to another. To be frank, that's just nonsense. Any DAW that uses 32 Bit floating point internal processing will be capable of processing audio at a quality where residual errors will be far below the limits of human auditory perception. We call this 'transparency'. That is, the ability to pass audio through the program unaltered in any way. Today, from a transparency perspective all DAWs are created equal and are able to play or process audio without audible effects caused by the 'audio engine'.

So where do these perceptions about differences in DAW quality come from, particularly since you see many 'famous' producers and recording engineers extolling the virtues of one DAW's audio quality or 24 Bit @ 96 kHz audio format? Surely these guys know what they are talking about. The answer is a little complex:

First, recording engineers and 'famous people' almost never assess the differences between DAWs or high-end audio formats under controlled conditions. That is, make statistically meaningful numbers of forced choice comparisons under double blind listening conditions, in controlled audio environments. What usually happens is that a company sends their product/s out free to as many industry famous people as they can find. This select group listens to it and a smaller sub-group think it "sounds better than anything they have ever heard". It is this sub-group and their quotes that appear in the marketing blurb for the product/s. Now don't get us wrong, these people really do think the product sounds great, but it's a subjective impression and hardly qualifies as proof that the product is any better than others in the market or even an improvement on what went before.

A second reason is that there are many settings and options that affect the live and rendered audio from any DAW. Its unlikely that 'out of the box' any two DAWs will make exactly the same sound. The following list will help you to understand what these settings and options are, and to give you a broader perspective on what really can make a difference and hopefully protect you from the marketing machine:

Live Mixer interpolation -

This applies to Sampler Channels when transposing samples from the root note. Plugin instruments may have their own live vs rendered interpolation settings. When sample-rates are converted (when pitch shifting a sample for example) then the DAW may need to call on sample data between the existing points. Interpolation is all about making an accurate prediction as to what that level should be, and so reduce a problem known as quantization errors and results in aliasing and or quantization noise.


Rendered audio settings -

Including WAV Bit-depth setting, MP3/OGG Bit-rate setting and Sampler interpolation. The wav Bit-depth (16, 24 or 32) won't really have much impact on the sound you hear however, the lossy formats (mp3 & ogg) definitely will introduce audible 'garbling' or 'underwater' sounds when used at Bit-rates less than about 190 kbps. These formats are really for music distribution, although can sound spectacularly good at Bit-rates of 240 kbps and above. Sampler interpolation is the same feature as discussed in the mixer section, but here applies to rendered files. If you are hearing differences between the live and rendered sound, then make sure the live and rendered interpolation settings match.

Your mixing decisions -

This is where the magic happens. If you can mix well, your music will probably sound great no matter what the technical specifications of the DAW are. Mixing is a craft and takes years to learn, just like any musical instrument. So if your mixes sound bad compared to the commercial mixes, this is, with 99.99% probability, the reason why. The DAW doesn't suck, you do. Also be aware that you don't need any more sophisticated tools to mix than a nice Parametric EQ, Compressor/Limiter and the basic Mixer functions. All those 'mastering' plugins are useful tools that can save time, but no substitute for experience and methodical work flow. If you want to get some idea how the actual sound itself can influence you emotionally, load for example the Vst Harmless in a default project and start working your way through the presets. Some sound thin and reedy others will blow your socks off. It's all about the sounds mixed together (the performance), not the technical specs of the DAW. Now imagine how hard it is to separate the performance from the technical aspects of the DAW when it comes to our emotional reaction to a platform. All too often, great mixing, performance or patch programming is mistaken for product design or specifications.

Loudness -

In a comparison, louder always sound 'better' than quieter. The louder of two otherwise identical sounds will seem to have more bass and clearer high frequencies. This comes from the way our ears work, not anything in the audio itself. You need to be very careful here, small level differences may not be apparent to you as 'louder' but as 'clearer', 'bassier' or 'crisper' as discussed earlier. As a rule-of-thumb, 1 dB is about the smallest level difference listeners can detect in a mix (~0.5 dB in a laboratory setting), so if you are comparing sounds they need to be matched to within 1 dB. Apart from basic mix decisions there are a number of reasons why sound rendered from FL Studio may be quieter/louder (depending on settings) when compared to the same audio/sample rendered from another DAW.

Plugins behaving differently -

Some plugins just sound bad or make strange noises when used with the wrong settings. This trips many people up when they render the same synth from two DAWs and compare the waveforms under a microscope. Synthesizers usually have some randomization and/or free running oscillators (meaning the phase of the waveform will change as a function of the note start time), as the point of most synths is to not produce the 'exact' same waveform twice. Make sure to disable any randomization settings and to send the same notes of the same velocity with any of the same modulation settings. A better strategy here is to use a .wav file as a test source, then you know it's identical to start with in each DAW.

Marketing has influenced you -

Yes it has. Digital audio is just a stream of numbers. Computers add up numbers in a well understood and predictable way, if they didn't we'd constantly have satellites raining down on us from the sky. We are talking here basic mathematics (addition, subtraction and division), there is no magic, there are no secret things that some DAW manufacturers know that others don't. Dithering and interpolation are well understood and there are plenty of options in most DAWs to give you control over them. But understand, that the suppliers of professional and consumer audio equipment have strongly vested interests in convincing you that you need to upgrade to the latest and greatest gear/format, that's how they make money, selling gear based on specifications. Audio quality ceased to be a meaningful differentiators of DAWs once they had all moved to 32 Bit float internal processing. The effect of a lifetimes marketing has been so powerful, let's consider three aspects associated with Bit-depth and sample rate: DAW Bit depth - Anyone who has used a calculator will know that when you perform mathematical operations you only have so many decimal places and so get rounding errors and these errors accumulate in the lest significant decimal place. The same thing happens when processing digital audio. If you have 16 Bit numbers representing your data then these rounding errors can creep into the audible realm, particularly with very quiet passages of music. A 32 Bit floating point format allows mathematical operations to be performed on audio without rounding errors becoming audible. Before you ask, no, 64 Bit is not subjectively better. Yes there are a few exceptional circumstances that can be concocted to create audible artifacts in a 32 Bit floating point format, but the same is true of 64 Bit float and these cases are not worth considering as a driving factor in perception of 'quality'.
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Old 08-11-2010, 09:37 AM   #2
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Default Part 2 :)

Electronics -

The best analog to digital electronic circuits that are currently possible to implement in commercially available 'professional' audio equipment are equivalent to, at best, 20 Bit. A dynamic range or 120dB. Yes, all those 24 Bit recordings are actually delivering somewhere between 18 to 20 Bits of real-world precision once they have been mangled by the best converters and room-temperature electronics you can buy. What this means is that even a lowly 24 Bit file has outstripped the ability of our electronics to reproduce it, the noise inherent in the electronic resistors and capacitors swamps the remaining resolution. Sample rates, on the other hand can go almost as high as you want, but as we have seen in the study above, more than 44.1 kHz is a waste. Time to stop worrying about the technical specifications of DAWs as a driver of 'quality' and concentrate on the other things in this list that matter more, like mixing and performance.

The weakest link - Human hearing. Surely 24 Bit 196 kHz wav files sound superior to the 16 Bit 44.1 kHz wav files used on CDs? Brace yourself, this may come as an unbelievable shock, the largest and best study conducted to date (see the reference below) shows that there is no audible difference between 'high-end' audio formats ~24 Bit @ 196 kHz and 16 Bit 44.1 kHz (CD standard). A user-friendly article discussing the research can be read at the following link: The Emperor's New Sampling Rate.

http://mixonline.com/recording/mixin...s_new_sampling

What they found -

From a sample of 554 listeners that included professionals, the general population & young listeners (prized for their high-frequency hearing), those that correctly identified the higher quality audio was 276, or 49.8%. The same number you would get if you just flipped a coin 554 times or asked untrained monkeys to do the task. In summary, 16 Bit 41.1 kHz sound is indistinguishable from ~24 Bit @ 196 kHz. Yes, 32 Bit float is important for audio processing in a DAW, but once it comes time to convert that audio into a format useful for distribution and human consumption, you just can't meaningfully improve on the CD standard.

You have influenced you -

You simply can't make an unbiased comparison of the audio from two sources, A and B, if you know what source you are listening to at any given time. You can't so forget it, perceptual psychologists realized this over 100 years ago (http://en.wikipedia.org/wiki/Psychophysics) and developed many useful methodologies to work around it. In particular the 'blind listening' experiment with an 'objective response indicator'. Get a friend to play you the two sources in random-order pairs. Your task is to simply identify source A and B (Two-alternative forced choice - Wikipedia, the free encyclopedia), nothing more, nothing less.

If you can distinguish source A vs B, at least 8 times, or more, from 10 random-order paired comparisons, then you may be able to hear something. If not, it is likely that you are just guessing. This is probably one of the most enlightening tests any audio-engineer can do, you will learn a lot about perception and your ability to hear things this way. Invariably we are much less sensitive than we think. As the sense of infallibility we have in our own perception is so strong, we have an exercise for you:

An experiment -

Render the same ~5 seconds of a project to a 320 kbps mp3 file and 16 Bit 44.1 kHz wav (CD format), then make 30 A vs B blind comparisons. This means you are not to know whether your helper is playing the mp3 or wav file to you. You should also avoid eye contact with them and receive no feedback on how well you are doing until the experiment is completed. The helper should write down a list of 30 comparisons randomizing the order to play them to you (wav vs mp3 or mp3 vs wav), they should ensure that you receive 15 mp3 vs wav and 15 wav vs mp3 trials, 30 in total in a mixed up sequence. You don't need to do them all at once, if you need a break do so, but don't confer with your helper about how you are doing. Your task is simply to identify the wav file (the better sounding one!). In order to convince a scientist, with a high degree of confidence you can tell them apart, you need to identify the wav file at least 20 times out of 30. Our untrained monkey fresh from his 24 Bit 96 kHz experiment will correct identify the wav file 15 times (chance). Surely, as mp3 is inferior to CD format, you can beat 20 correct identifications or at least the untrained monkey?

Your soundcard or Windows mixer & Media player -

Make sure you don't have any EAX, Compressors or EQ settings etc., on in the soundcard or Windows settings. Dig deep here, sometimes they are well hidden in 'advanced' options tabs etc.

Some time in the late 1990's we moved beyond the point where technological improvements to 'fidelity' ceased to make any meaningful difference to the audible quality of the music we produce. Further, with the Loudness Wars (http://en.wikipedia.org/wiki/Loudness_wars) of the 2000's and widespread adoption of low Bit-rate mp3's as a music distribution standard, it's clear that audio quality has been going backwards for a while now, but people are still enjoying their music.

In conclusion, we'd like to leave you with a quote from photographer Vernon Trent -

"Amateurs worry about equipment, professionals worry about money, masters worry about light. I just take pictures"


References

Meyer, E. Brad and David R. Moran. Audibility of a CD-Standard A/D/A Loop Inserted into a High-Resolution Audio Playback, Journal of the Audio Engineering Society, Sept. 2007, pp. 775-779.

Audio Myths Workshop, Audio Engineering Society 2009

Finally, when you have an hour to spare one night, here's a video well worth watching, Ethan & co-presenters cover many of the issues discussed above, including placebo effects in audio, loudness vs quality, 'scam' equipment, dithering, expensive vs cheap soundcards and more...

Ethan Winer's, YouTube video, Audio Myths Workshop AES 2009

https://www.youtube.com/watch?v=BYTlN...layer_embedded


Article by Image Line Team
http://www.image-line.com
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Old 08-11-2010, 09:47 AM   #3
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47.8% of statistics are made up.

If you want to start a discussion about what people hear, you should read the recent results of a test I ran on the blog... irrelevant of what the science says it's what people hear that really matters http://reaperblog.squarespace.com/ho...e-results.html

PS: The stats in the test are 'real'!
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Old 08-11-2010, 09:50 AM   #4
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THANK YOU Norbury. Now let's go record some tunes.
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Old 08-11-2010, 09:53 AM   #5
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Brilliant post. Thanks!
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Old 08-11-2010, 10:07 AM   #6
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Quote:
Originally Posted by norbury brook View Post
Here's a little something for you all

Audio Myths & DAW Wars

etc etc
interesting read
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Old 08-11-2010, 10:41 AM   #7
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Quote:
Originally Posted by RussUK View Post
47.8% of statistics are made up.
... but because of that in itself being a statistic, only 87,4% of that statement is true




Funny that, I thought I typed a random number and only after that realized it was the same number backwards
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Old 08-11-2010, 10:42 AM   #8
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For once a post about this useless debate that does not aim at just stiring up some sh*t It's refreshing.

When I am on stage, I could be playing my 3000$ Gibson Les Paul or my 1500$ Carvin DC400W or my 600$ Ibanez S470DXQM ... people just don't give a crap and none of them could hear the difference, and to be honest, when on stage with the crowd, the noise, the place's accoustic and the other instruments...if it wasn't from the "in ear monitors", I would not also. In the end, my expensive guitars are for "me", and as musicians, artists and humans we develop our habits, superstitions, preconceptions and delusions. Shrinks and Doctors use and believe in the placebo effect for a reason, it's true and it works. But my Carvin really does sound better than my other guitars

Let's be honest, if DAW X did not sound better than Y, and X cost 10 times the price of Y, and we did pay the caboodle for X...would we really and openly admit that X is the same?...as humans, we would not because we would try to convice ourselves we did not get screwed or we would firmly believe it's better because we are "partial" to X a number of reasons. This is just being human.
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Old 08-11-2010, 10:46 AM   #9
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Audio sounds better if you put blue highlighter on your tweeter every other Thursday, unless it falls on a calendar day(Mayan based of course) with a 3 in it it, then you use yellow until the next non-3 day.
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Old 08-11-2010, 10:47 AM   #10
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Thanks for that. Interesting read.
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Old 08-11-2010, 10:48 AM   #11
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The "sound" of a DAW is a silly argument as has been stated. First, what does "better" mean. And really, shouldn't you have the skills to mix the tune to sound how you want it? Any difference in DAW sound will be relative and you should be able to adjust to it simply by listening.
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Old 08-11-2010, 10:52 AM   #12
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Originally Posted by norbury brook View Post
Research has shown that for music distribution, 16 Bit @ 44.1 kHz (CD standard) is indistinguishable from 24 Bit @ 196 kHz in a sample of over 550 listeners. In other words, more Bits and higher Bit-rates are not going to improve the quality of your tracks.
There is one huge advantage in higher sample rate: less frequency band mirroring, knows as Nyquist-Shannon theorem.

You all probably know that, but some newbies don't: a lot of non-linear effects (distortion, compressors, etc.) introduce higher harmonic components and if you record and apply effects in 44.1kHz, you can have a lot of frequency band mirroring and no filtering could help. Only smart and interpolating (oversampling) effects can eliminate mirroring. But if you record in 192kHz, this mirroring is almost irrelevant.

But I agree: there is no hearing difference between them. So it is best to record and perform effects in 192kHz, but at the end, downsample the track to 44.1kHz.
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Old 08-11-2010, 10:54 AM   #13
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Quote:
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You all probably know that, but some newbies don't: a lot of non-linear effects (distortion, compressors, etc.) introduce higher harmonic components and if you record and apply effects in 44.1kHz, you can have a lot of frequency band mirroring and no filtering could help. Only smart and interpolating (oversampling) effects can eliminate mirroring. But if you record in 192kHz, this mirroring is almost irrelevant.
There's a better solution than writing a lot of zeros in contrast to 48k/24-bit. Just use effects which have internal oversampling and you don't really need to use 192k then. Anyways you can always use a steep lowpass filter with high cutoff on the track after the offending effect.
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Old 08-11-2010, 10:59 AM   #14
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great read thanks
I love threads like this one

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Old 08-11-2010, 11:00 AM   #15
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Quote:
Originally Posted by EvilDragon View Post
There's a better solution than writing a lot of zeros in contrast to 48k/24-bit. Just use effects which have internal oversampling and you don't really need to use 192k then. Anyways you can always use a steep lowpass filter with high cutoff on the track after the offending effect.
I have a lot of VST, but almost no-one has documented that it uses oversampling So as a pessimist, I rather use 192kHz :P

Steep lowpass filter can only help to reduce the 22kHz-44kHz part. But mirroring occur also in 0-22kHz. Let say we have a pure sine at 5kHz. The distortion adds 15kHz, 25kHz, 35kHz, 45kHz, 55kHz...
25kHz gets mirrored into 19kHz, 35 into 9kHz, 45kHz into 1kHz, etc. So how do you use lowpass here? You can only trust your VST to be oversampled. But if you don't trust them, you use higher sampling rate.
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Old 08-11-2010, 11:08 AM   #16
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Thin notch filters and highpass for "the other side" also help


But yeah, better is to use oversampled and/or band-limited effects and instruments. A lot of VSTi nowadays have oversampling and/or built in (rgc:audio and Cakewalk synths, ImageLine synths, TAL synths, u-he, Spectrasonics, NI, etc.) And sometimes you just don't care and let 'em have their fizzy aliasing!
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Old 08-11-2010, 11:08 AM   #17
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The audio quality of a DAW, converters, sample rates and bit-depths... is just a distraction to keep us from discussing how boring, overproduced, clinically processed to sterility, uninspired, repetitive, heard-this-a-million-times-before... music is nowadays.
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Old 08-11-2010, 11:09 AM   #18
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Quote:
Originally Posted by jmcecil View Post
Audio sounds better if you put blue highlighter on your tweeter every other Thursday, unless it falls on a calendar day(Mayan based of course) with a 3 in it it, then you use yellow until the next non-3 day.
i was told to use green, and then yellow.
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Old 08-11-2010, 11:18 AM   #19
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Quote:
Originally Posted by EvilDragon View Post
Thin notch filters and highpass for "the other side" also help


But yeah, better is to use oversampled and/or band-limited effects and instruments. A lot of VSTi nowadays have oversampling and/or built in (rgc:audio and Cakewalk synths, ImageLine synths, TAL synths, u-he, Spectrasonics, NI, etc.) And sometimes you just don't care and let 'em have their fizzy aliasing!
Yep, we don't care, as fortunately, in "controlled environment" they don't have much effect. Normally we only use slight distortions (warmer, exciter), and many other sounds which masks the aliasing.

But if you only have one instrument and apply distortion with using Reaper's plugins, you get what I have attached here.
Attached Images
File Type: jpg nyquist.jpg (66.2 KB, 453 views)
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Old 08-11-2010, 11:24 AM   #20
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...and that's why we don't use those JS distortions. :P At least not over here.
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Old 08-11-2010, 11:27 AM   #21
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...and that's why we don't use those JS distortions. :P At least not over here.
Of course, that was just an over-exaggerated example to warn newbies

Still I suppose a lot of people use some of JS plugins, otherwise they won't be there. I am aware of aliasing, so I use plugins carefully and check if they are oversampling.
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Old 08-11-2010, 11:29 AM   #22
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I'll preface this by stating that I completely agree with Norbury's posts. It's great to see these things presented in such a concise manner.

However, as stated already some digital non-linear processors exhibit aliasing (this is completely avoided by oversampling - you don't have to oversample to 192kHz!! that's *overkill*).

Also, some EQs have HF curves that will sound better/different when oversampled, because if not, the curve changes shape and becomes constricted as the centre-frequency approaches sr/2. Using these eqs at a higher sample rate, or engaging oversampling if supported completely avoids this issue, and again, going as high as 192kHz is a massive waste of cpu resources. Or simply use a good LPF as EvilDragon suggested.

Great thread.
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Old 08-11-2010, 11:31 AM   #23
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I was going to add the same thing Tim, if I'm not working at 96k I ALWAYS oversample when I can


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Old 08-11-2010, 11:43 AM   #24
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I must say (as I am aware of the aliasing and am careful with the plugins) that in fact I only use 44.1kHz recording. There are a lot of reasons for that(CPU and disk are not, have powerful computer).

But I wanted to say, that using higher sampling rate is not so stupid and useless, if only you know why.
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Old 08-11-2010, 11:51 AM   #25
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Originally Posted by norbury brook View Post
"You have influenced you"
I knew I couldn't trust myself!

I don't think there was anything in parts 1 & 2 that I vehemently disagreed with. It was a good read.

JR
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Old 08-11-2010, 11:55 AM   #26
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Great read NB. Apply it to everything else, like MP3's. I posted a link to that vid some months ago...

http://forum.cockos.com/showthread.p...highlight=myth

I recall a big fuss about that workshop on another forum even before it happened. Ethan announced the intent of doing this and was basically attacked...

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Old 08-11-2010, 12:20 PM   #27
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Quote:
An experiment -

Render the same ~5 seconds of a project to a 320 kbps mp3 file and 16 Bit 44.1 kHz wav (CD format), then make 30 A vs B blind comparisons. This means you are not to know whether your helper is playing the mp3 or wav file to you....
HydrogenAudio.org has links to some software for automating blind-ABX tests. (You don't need a helper... just feed-in the two files and tell the computer which one you think you're hearing.)

With high-bitrate lossy compression, it turns-out that the biggest factor is program material. With most music you won't hear a difference, but transients (castenets are the classic example) create an artifact called "pre-echo" in MP3s, and if you know what to listen for you can identify the MP3.

P.S.
I'm NOT advocating the use of MP3s in audio production. It just helps to put things in perspective... If an MP3 can be transparent, we shouldn't get too hung-up about using the highest-possible sample-rates & bit depths.

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Old 08-11-2010, 12:24 PM   #28
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Another possible confusion point here is that we are talking the final playback item. Not, individual tracks of a project. As long as the recording is solid and the tracks don't go through multiples of bit depth conversions you should be ok. But, continuous truncation by plugins and the DAW engine IS audible. The danger of 16 bit media IN a project is constant truncation. If you let it pile up it will be audible. The same is true for higher bit depth media, but it takes longer for the problems to show up.

But, in principle I agree. I've heard CDs that were breathtaking and DVD-As that sound like crap. Unfortunately for me, my skills at mixing are aligned with those guys that make DVD-A sound like crap.
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Old 08-11-2010, 12:34 PM   #29
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Another possible confusion point here is that we are talking the final playback item. Not, individual tracks of a project. As long as the recording is solid and the tracks don't go through multiples of bit depth conversions you should be ok. But, continuous truncation by plugins and the DAW engine IS audible. The danger of 16 bit media IN a project is constant truncation. If you let it pile up it will be audible. The same is true for higher bit depth media, but it takes longer for the problems to show up.

But, in principle I agree. I've heard CDs that were breathtaking and DVD-As that sound like crap. Unfortunately for me, my skills at mixing are aligned with those guys that make DVD-A sound like crap.

yes definitely agree 100% with that


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Old 08-11-2010, 03:14 PM   #30
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There's an ABX JS plugin available in Reaper (as well as an ABCDX one). I sure hope we take away the approach to evaluating our tools and methods that norbury suggests as well as his results. Honestly answer that first question "can I hear a difference at all?" and we'll always be confident in our opinions.
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Old 08-11-2010, 03:52 PM   #31
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You are underestimating the hearing of untrained monkeys. Studies have shown that many monkey species are able to hear one octave or more than humans, so hearing up to 45kHz would make a difference when sampling at 96kHz.
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Old 08-11-2010, 05:15 PM   #32
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So far I've never been asked to mix for the monkey demographic.

EDIT: However, if a sizable simian community suddenly finds itself with enough discretionary income to become regular music purchasers, I wouldn't be apposed to doing so.

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Old 08-11-2010, 07:18 PM   #33
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Very interesting post, I've only one thing to add.

Quote:
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Research has shown that for music distribution, 16 Bit @ 44.1 kHz (CD standard) is indistinguishable from 24 Bit @ 196 kHz in a sample of over 550 listeners.
The above may well be true, but I'd suggest that what follows may be a non-sequitur.

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In other words, more Bits and higher Bit-rates are not going to improve the quality of your tracks.
Au contraire (for me at least,) recording at 24 bit brings with it the significant advantage of more headroom, making it easier to avoid clipping. This makes it easier to obtain a good clean recording, especially when you are dealing with a vocalist who has the tendency to scream from time to time.

The mix of course gets rendered down as 16 bit eventually anyway ...
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Old 08-11-2010, 07:57 PM   #34
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recording at 24 bit brings with it the significant advantage of more headroom, making it easier to avoid clipping. This makes it easier to obtain a good clean recording, especially when you are dealing with a vocalist who has the tendency to scream from time to time.
The mix of course gets rendered down as 16 bit eventually anyway ...
+100 to that point. 16 bits can get a bit long in the tooth for recording. For example if there happens a case where the signal needs to be boosted a lot to make it audible. A lot meaning something like 40 decibels. In that case the low resolution of the 16 bit signal could very well be exposed in an audible way. Something like that is of course a bit pathological case, but if using 24 bits resolution while recording avoids or at least alleviates such possible problems, there's no reason not to record at 24 bits. Also for internal DSP within the DAW and plugins, 32 or even 64 bit floating point isn't really overkill. There are cases when the additional available signal range can be very useful. And it doesn't really even cost that much in terms of CPU load as CPUs calculate floating point numbers quickly anyway.
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Old 08-11-2010, 08:02 PM   #35
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+100 to that point. 16 bits can get a bit long in the tooth for recording. For example if there happens a case where the signal needs to be boosted a lot to make it audible. A lot meaning something like 40 decibels. In that case the low resolution of the 16 bit signal could very well be exposed in an audible way. Something like that is of course a bit pathological case, but if using 24 bits resolution while recording avoids or at least alleviates such possible problems, there's no reason not to record at 24 bits. Also for internal DSP within the DAW and plugins, 32 or even 64 bit floating point isn't really overkill. There are cases when the additional available signal range can be very useful. And it doesn't really even cost that much in terms of CPU load as CPUs calculate floating point numbers quickly anyway.
+++ numbers

The difference between 32 and 64-bit fp are pretty extreme if you start playing around with varispeed and suchlike........most processing actually.
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Old 08-11-2010, 09:49 PM   #36
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Originally Posted by nicholas View Post
Au contraire (for me at least,) recording at 24 bit brings with it the significant advantage of more headroom, making it easier to avoid clipping. This makes it easier to obtain a good clean recording, especially when you are dealing with a vocalist who has the tendency to scream from time to time.

The mix of course gets rendered down as 16 bit eventually anyway ...
Quote:
Originally Posted by norbury brook View Post
Research has shown that for music distribution, 16 Bit @ 44.1 kHz (CD standard) is indistinguishable from 24 Bit @ 196 kHz in a sample of over 550 listeners. In other words, more Bits and higher Bit-rates are not going to improve the quality of your tracks.
The whole first paragraph is referring to distribution, NOT actual tracking.
Recording in 24 bits is the way to go.
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Old 08-11-2010, 11:23 PM   #37
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Very interesting post, I've only one thing to add.



The above may well be true, but I'd suggest that what follows may be a non-sequitur.



Au contraire (for me at least,) recording at 24 bit brings with it the significant advantage of more headroom, making it easier to avoid clipping. This makes it easier to obtain a good clean recording, especially when you are dealing with a vocalist who has the tendency to scream from time to time.

The mix of course gets rendered down as 16 bit eventually anyway ...
Look, if 16 bit is all you have - make music. If an old Tascam four track is all you have - make music. I agree completely with the above and would like to add the obvious - you achieve better VST results with audio recorded at ~24 - I don't even think this is debatable - which is pretty much the only reason I'm an avid believer in 24 over 16.
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Old 08-11-2010, 11:28 PM   #38
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47.8% of statistics are made up.
I hope you don't mind if I steal that for my signature Pretty please? lol

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Old 08-11-2010, 11:46 PM   #39
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I hope you don't mind if I steal that for my signature Pretty please? lol

Ben
Steal away - 100% of jokes are someone else's!

I'm also currently doing some research on the effect of eating different cheeses on audio compression. So far the french soft cheeses are perfect for vocal tracking whilst a hard English strong cheddar seems to be the right way to go for Rock drums. Anyway more to go, I'm struggling with a pre-grated Italian parmesan for vocals, it seems to give vocals a sibilant side effect.
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Old 08-12-2010, 12:19 AM   #40
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Steal away - 100% of jokes are someone else's!

I'm also currently doing some research on the effect of eating different cheeses on audio compression. So far the french soft cheeses are perfect for vocal tracking whilst a hard English strong cheddar seems to be the right way to go for Rock drums. Anyway more to go, I'm struggling with a pre-grated Italian parmesan for vocals, it seems to give vocals a sibilant side effect.
A nice aged blue cheese goes a long way to correct phase issues is what I last heard on the streets
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