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I just use mah ears. :)

ABX testing is your ears. :P

 

It just hides the tracks under the labels 'A' and 'B' and asks you to select the one which sounds best, so you can't influence the result through bias.

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Guest xyzzy frobozz

I just use mah ears. :)

ABX testing is your ears. :P

 

It just hides the tracks under the labels 'A' and 'B' and asks you to select the one which sounds best, so you can't influence the result through bias.

 

I'm someone who can hear and cannot stand "slammed" recordings of the loudness wars, let alone low bit rate conversions.

 

Even on some very high end equipment with expensive headphones, I can't hear the difference between 320mp3 and lossless. I personally doubt that anyone can discern modulations at 1/320th of a second....

 

At that rate you're much more likely to be hearing idiosyncrasies of the recording equipment than the recording itself.

Edited by xyzzy frobozz

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I'm someone who can hear and cannot stand "slammed" recordings of the loudness wars, let alone low bit rate conversions.

 

Even on some very high end equipment with expensive headphones, I can't hear the difference between 320mp3 and lossless. I personally doubt that anyone can discern modulations at 1/320th of a second....

 

At that rate you're much more likely to be hearing idiosyncrasies of the recording equipment than the recording itself.

Yep, the problem with music these days isn't the codec, it's how it's mastered. Dynamic range compression is the bane of music IMO.

Edited by .:Cyb3rGlitch:.

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I just use mah ears. :)

ABX testing is your ears. :P

 

It just hides the tracks under the labels 'A' and 'B' and asks you to select the one which sounds best, so you can't influence the result through bias.

 

 

 

I knew that.......

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You won't hear the difference between a good high bitrate lossy codec over a lossless one. 320kbps MP3 is perfectly sufficient - I certainly cannot honestly tell the difference with A-B testing on my planar magnetic headphones + Essence STX.

Silly people spend so much time going over the differences between codecs and their psychoacoustic models that they train themselves to listen to the deficiencies of each (pre-echo and etc) on a very specific and limited selection of samples.

 

Which is a cool party trick, but way to ruin your listening experience :

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You won't hear the difference between a good high bitrate lossy codec over a lossless one. 320kbps MP3 is perfectly sufficient - I certainly cannot honestly tell the difference with A-B testing on my planar magnetic headphones + Essence STX.

Silly people spend so much time going over the differences between codecs and their psychoacoustic models that they train themselves to listen to the deficiencies of each (pre-echo and etc) on a very specific and limited selection of samples.

 

Which is a cool party trick, but way to ruin your listening experience :

 

For sure, I only ran the tests on a few of my favourite tracks, and didn't hear a difference. That's good enough for me.

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Even on some very high end equipment with expensive headphones, I can't hear the difference between 320mp3 and lossless. I personally doubt that anyone can discern modulations at 1/320th of a second....

excepting the fact there is no correlation like the one you appear to imply — between bitrate and audibility of oscillations (modulated or otherwise) — we agree.

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They are pretty amazing.

At first I often thought someone was at/opening my door, but it was just background sounds in music.

I've turned up the 4/8/16k frequencys up a tad (linear). Sounds much better, but that's likely my speakers.

Edited by mudjimba

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A few technical points to clear up some interesting ideas ie "discern modulations at 1/320th of a second".

 

Number of bits (ie 16bit for CD's) - The "size" of a sample, this affects the dynamic range (the difference between the loudest and quietest level) that can be represented. 16bit is more than enough to handle the range in practically all popular music. 24bit is enough range to cover the range from below the level the human ear is capable of hearing to well beyond the threshold of pain. Movies generally have a much wider dynamic range and can benefit from 20 or 24bit. 24bit is also common with recording as it allows more headroom to capture sudden peaks without clipping.

 

Sample rate (ie 44Khz for CD's) - This number is the number of samples per second. Differences in sample rate can be more obvious but again nobody can really tell the difference between 44Khz and higher in A-B tests. In many cases lowering the sample rate a little is similarly hard to notice (there are plugins for that let you do this in real time). Again when recording higher sample rates are used to provide more resolution when applying plugins and mixing multiple tracks.

 

Bit rate - For uncompressed files (ie WAV or just CD Audio) bit rate is simply the number of bits being processed per second, for uncompressed music this is easy:

bits-per-sample x sample-rate x number-of-channels for a (standard) CD 44100 × 16 × 2 = 1411200 bits per second = 1411.2 kbit/s

For compressed files this is different (or variable in the case of VBR) in that it is the number of bits per second feed into the codec, not as the quote at the beginning of this post states. Technically I could create my own implementation of MP3 that pads each MP3 frame with some extra 0's so as to increase the bit rate to that of the PCM encoding used on CD's, it would be a pointless implementation but it would technically have the same bit rate (actually MP3 already does this to maintain a consistant bit rate).

 

And sorry .:Cyb3rGlitch:. while the loudness war ended a while ago and dynamic range compression is not the bane of music, it is an important tool for levelling a recording. Compression is used at many stages during the recording/mixing process to smooth out a performance (even the best musicians on the planet will still have peaks where they hit a string or drum a bit harder). What you are referring to is the late 90's technique of applying a heavy compression to the final master to squeeze the audio into the top few db just below (or in some cases right up to) the point were clipping occurs, this was to try and make the recording "sound" louder as a louder track "sounds better". Thankfully sanity has prevailed and we are back to getting recordings released with a couple db of headroom available and a reasonable amount of dynamic range.

Edited by SledgY

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yes, by and large some degree of compression is warranted/necessary at most stages of audio production.

 

i agree with everything above Sledgy, except for your claim that "16bit is more than enough to handle the range in practically all popular music". i would say that 20 bits are more than enough for any purpose, but 16 still leaves a sizeable noise floor. whilst you could certainly argue that most 'popular music' does not make extensive use of dynamic range and therefore does not tend to expose the differences between 24 and 16 as readily as other genres, i think it is far too broad a category to make such a pronouncement (even with your qualifier "practically all" duly noted).

Edited by @~thehung

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yes, by and large some degree of compression is warranted/necessary at most stages of audio production.

There is a little more than some, compression is an essential tool (one of the main tools along with EQ) in audio production. Compression is also fairly simple concept, to put it plainly (ie ignoring the more advanced parameters), any signal above certain threshold has gain reduction applied to the output signal. The amount of gain reduction applied is determined by a ratio, ie if the input signal is 4db above the threshold and the ratio is set to 1:4, a gain reduction is applied which produces an output that is only 1db above the threshold. The origins of the audio compressor is from the early days of recording when an operator would manually adjust the gain control to keep the output level consistant.

 

Certain instruments (bass, kick and snare) often have compression applied. Drums as a whole typically have compression (sometimes a lot) applied in multiple stages, on each track and on the entire drum bus.

 

i agree with everything above Sledgy, except for your claim that "16bit is more than enough to handle the range in practically all popular music". i would say that 20 bits are more than enough for any purpose, but 16 still leaves a sizeable noise floor. whilst you could certainly argue that most 'popular music' does not make extensive use of dynamic range and therefore does not tend to expose the differences between 24 and 16 as readily as other genres, i think it is far too broad a category to make such a pronouncement (even with your qualifier "practically all" duly noted).

As for comments regarding 16bit I am confident in my statement that 16bit is capable of pretty much all music (I say pretty much to exclude examples of silly outliers like noise bands and the like). 16bit gives a theoretical 96db of dynamic range, with a typical noise floor at around -90db for even the nastiest audio system and typical background noise/ambient noise being greater than this level 16bit is enough for most recorded music. Music typically does not have a wide dynamic range. Don't believe me? Get a recent recording and open it up in your favourite audio editor (audacity is fine). Chances are the waveform wil be fairly consistently at the same level and will not being using much of the available range, remember the width of the waveform (or amplitude) is directly related to the volume and the distance between the highest level and the lowest level being the Dynamic Range.

 

/edit This guy explains it even better, http://www.head-fi.org/t/415361/24bit-vs-1...e-myth-exploded his post is right on the money.

 

One of the hardest things about audio is that there is a lot of cognitive bias that comes into play, if you are trying to hear a particular sound or you are expecting to hear a particular sound then you are more likely to do so. A good example of this is the demo of Stairway to Heaven being played backwards, if you just listen to it funnily enough it sounds like a record being played backwards, but if your are then given an example of lyrics supposedly praising satan and listen again all of a sudden the backwards lyric sounds a lot like somebody praising satan. Also similar is the effect the volume has, a song played loud just sounds better, this is to do with the way sound is interpreted by your ears. In the end the goal is always something that sounds good but changing from 16bit to 20bit is an example of something that doesn't have a great outcome, switching to better speakers or improving the acoustics of your room will give you a much better improvement in audio quality.

Edited by SledgY

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While that's an interesting enough read, logical fallacy alert:

 

2 = The concept of the perfect measurement or of recreating a waveform perfectly may seem like marketing hype. However, in this case it is not. It is in fact the fundamental tenet of the Nyquist-Shannon Sampling Theorem on which the very existence and invention of digital audio is based. From WIKI: “In essence the theorem shows that an analog signal that has been sampled can be perfectly reconstructed from the samples”. I know there will be some who will disagree with this idea, unfortunately, disagreement is NOT an option. This theorem hasn't been invented to explain how digital audio works, it's the other way around. Digital Audio was invented from the theorem, if you don't believe the theorem then you can't believe in digital audio either!!

Just because DA was developed out of that specific theory, it doesn't then mean that the theory must be correct. Otherwise, every paradigm shift in science, we'd all be pretty fucked!

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Just because DA was developed out of that specific theory, it doesn't then mean that the theory must be correct.

It would be more accurate to say that the theory lead to the development of digital audio by mathematically proving that a signal could be recreated such that it is a copy of the original signal. He is trying to point out that the theory came first and that the theory proved that digital audio was possible.

 

Lets add one more this time from the guys over at xiph.org, if anybody knows a thing or two about digital audio it's these guys (being responsable for org/vorbis and FLAC). The article is primarily about how 96khz/24bit audio but covers the other issues well, it is also very well cross referenced to many other good resources on the topic. http://people.xiph.org/~xiphmont/demo/neil-young.html

 

If you really want to become completely informed on this topic and not parrot the same missinformation I very much recommend that article.

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true enough.

 

i was factoring in the recording end of things, where 16 is not enough. at the playback end, i dont suppose i could have ever heard dynamic range differences, all things being equal. however, ive long favoured the idea of a 24-bit/48kHz release format because it simplifies some of the mastering. this is probably from having heard my share of 16-bit versions of 24-bit masters that sound somewhat degraded. granted, this is probably due to noise rather than anything else, but when it is most noticeable as clean crisp sounds fade to black in pristine listening conditions, it is tantamount to a loss of 'range'.

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I'm not sure how 24bit/48khz simplifies mastering... A master is produced from the final 24bit/48Khz mix, the and the level normalised. At this point we are looking at dynamic range of ~12db max, after downsampling to 44Khz we are left with a CD quality stereo track with the same dynamic range, it just takes up less storage. If a 16bit master sounds bad it's because the master is bad not because it's 16bit.

 

24bit in recording is largely for headroom reasons, you can recording perfectly fine at 16bit provided you use compression before you hit the DAC to prevent clipping. The extra headroom allows you to record just the raw input and apply compression afterwards, letting you play with the settings more later.

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"24bit in recording is largely for headroom reasons"

 

um, no. i think you will find it is for added sparkle. headroom? never heard of it.

 

16-bit + compressor is not "perfectly fine" if you dont want to introduce errors/noise and interfere with linearity. hence, the next 4 useful bits above 16 = pure unadulterated sparkle.

 

as for differences between 16 and 24bit masters — they happen, in practice, when digitial gear working in real life signal chains is pushed beyond some limit that does not even exist on paper. they dont have to happen, and shouldnt happen, but they do. and its something that may not always be written off as the result of cheap gear, small internal word sizes, flaky dithering plugins, or operator error. and it all means the same thing = a loss of pure unadulterated sparkle.

Edited by @~thehung

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"24bit in recording is largely for headroom reasons"

 

um, no. i think you will find it is for added sparkle. headroom? never heard of it.

 

16-bit + compressor is not "perfectly fine" if you dont want to introduce errors/noise and interfere with linearity. hence, the next 4 useful bits above 16 = pure unadulterated sparkle.

 

as for differences between 16 and 24bit masters — they happen, in practice, when digitial gear working in real life signal chains is pushed beyond some limit that does not even exist on paper. they dont have to happen, and shouldnt happen, but they do. and its something that may not always be written off as the result of cheap gear, small internal word sizes, flaky dithering plugins, or operator error. and it all means the same thing = a loss of pure unadulterated sparkle.

I take it then you do not have experience with recording.

 

Headroom: When you are recording you set the input gain on your microphone preamp (or line input) so that the average level sits around the middle (or a little above) on the meters. This means that the difference between that midpoint and the maximum level (before clipping occurs). Obviously here a 24bit sample size has a much wider range and hence headroom. The wikipedia info

 

Sparkle: When somebody asks for sparkle it's usually involves tweaking the EQ on the high end frequencies, nothing to do with bit depth.

 

All of the masters I've had done are from 24bit stems/mixes. Once the master is completed it is then down-sampled to 16bit/44Khz. Digital gear does not get pushed beyond it's limits, one hard limit on digital gear is the peak level, you can't go past it, hitting that limit is called digital clipping, it sounds crap you don't go there, sorry man but you are starting enter into audiophile fantasy land territory.

Edited by SledgY

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haha you passed the aspy test.

Heh spent a lot of time researching digital audio, have an ever expanding rack of gear out of it. Don't start a studio, is one deep money pit! Although a lot cheaper for the band than paying studio rates!

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actually, i was referring to you not properly inferring my tone.

 

the fact i entered the thread to offer a technical correction of sorts mightve been some indication i wasnt in need of being talked at on the topic of compression. nevertheless, i was quite happy to let that (and successive lectures) slide, for their potential benefit to others. i also conceded the point about dynamic range at the playback end (with some reservations).

 

but i think you need to read with a little more grace and a little less presumption. the fact you could actually believe i was serious about "pure unadulterated sparkle" or not having heard of headroom... (*gobsmacked*)

 

having considered this further, particularly in reference to operas ive recorded, nailing down all possible sources of the degradation is academic. because you see, i want to be able to preserve everything from booming timpani capable of shaking windows, to a distant triangle trailing off into nothing but room tone and component self-noise — without the introduction of audible quantisation error and dither.

 

this is true even for sounds falling below the threshold of audibility under ordinary listening conditions. it is worth keeping in mind at this point that an opera can meander to a sustained crescendo over the course of 3 or more hours. if a listener likes a particular quiet section, that might end up as a track destined for aggressive normalisation at the hands of their ripping software and/or just purposively listened to very loud. "too loud" one could argue — but if there is extra detail coming from a closely miked choir which would otherwise be denied them (due to the relatively distant theoretical listening position of the mix), why shouldnt they hear it with the utmost clarity?

 

i have perceived the difference between 24 and 16 in ways extending from the very subtle (and no doubt at least bordering on self-deception) to the unsubtle. the latter has been when mixing and routinely dealing with whispers amplified to the level of shouts. in that respect, with a super clean signal, 24-bit audio is the gift that keeps on giving. the utility it offers is akin to zooming in on a photo with a resolution beyond what is required for orthodox printing or display. in the end, it is not for you or i to scoff at listeners who would appreciate having the option to defy the ordinary 'rules' of listening in a similar way, for whatever reason.

 

a release medium should cover all bases. 16 bit is enough, with some extremely marginal qualifications. 24 bit is more than enough, period. in most practical respects those extra bits are excessive, but i would like to think improvements in the economy of digital storage are increasingly making any potential of diminished returns a non-issue.

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actually, i was referring to you not properly inferring my tone.

 

the fact i entered the thread to offer a technical correction of sorts mightve been some indication i wasnt in need of being talked at on the topic of compression. nevertheless, i was quite happy to let that (and successive lectures) slide, for their potential benefit to others. i also conceded the point about dynamic range at the playback end (with some reservations).

 

but i think you need to read with a little more grace and a little less presumption. the fact you could actually believe i was serious about "pure unadulterated sparkle" or not having heard of headroom... (*gobsmacked*)

Tone is not obvious to others a text based medium, part of my recent role was to deliver technical lectures and it's something I shift into, although a little bit of trolling on your part ;). I like to ensure that facts are represented and a little confrontation is not always a bad thing (provided it doesn't move into personal attacks). My comments regarding compression where more aimed at missinformation surrounding the "90's noise wars".

 

Operas/Classical are not something I've recorded myself and am all ears on this topic my early musical training was in a 30+ brass band primarily on percussion, am very familier with the massive sounds you can get from timpani! Being a percussionist/drummer I have spent most of my time going after a great drum sound.

 

Anyway great to know there are others on these boards with similar interests/experiences.

 

- Tim

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Do you guys know if Dre beats solo HD that I own, would be worth the purchase of his $170 card? Would the headphones be significantly improved making this purchase worthy?

 

Dre beats solo HD

Xonar essence STX

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