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#21 Flouncy

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Posted 08 January 2013 - 01:34 PM

Kate Bush is one classy clever beautiful lady with magical amazing songs and a voice you could hug. Lossless... just lossless. Dunno. When I ripped CDs (now all done, haven't done it in years), I'd select lossless for the stuff I really wanted max fidelity on. So yeah I guess it is 24bit. But dunno. Not an expert at this.
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#22 fajw

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Posted 08 January 2013 - 01:57 PM

Normal CDs are not HD; they are 16 bit @ 44.1 kHz per channel.

#23 .:Cyb3rGlitch:.

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Posted 08 January 2013 - 02:29 PM

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.

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#24 Director

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Posted 08 January 2013 - 02:42 PM

Yeah it's funny, most of the time I reckon my hearing is crap but I can tell the difference between flac and MP3, and the above mentioned 24 tracks from Kate seem a lot richer too?

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#25 .:Cyb3rGlitch:.

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Posted 08 January 2013 - 02:47 PM

I haven't got that album to try for myself. How I usually test is to get a lossless copy, convert it to 320kbps MP3 using Audacity, and then do an ABX test using foobar2000.

Edited by .:Cyb3rGlitch:., 08 January 2013 - 02:51 PM.

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#26 Director

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Posted 08 January 2013 - 03:27 PM

I just use mah ears. :)

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#27 .:Cyb3rGlitch:.

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Posted 08 January 2013 - 03:38 PM

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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#28 Guest_xyzzy frobozz_*

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Posted 08 January 2013 - 07:42 PM

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, 08 January 2013 - 07:45 PM.


#29 .:Cyb3rGlitch:.

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Posted 08 January 2013 - 07:58 PM

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:., 08 January 2013 - 07:59 PM.

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#30 Director

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Posted 09 January 2013 - 07:50 AM

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.......

"The most powerful tool in the hand of the opressor is the mind of the opressed."-- Steve Biko

Those Who Dance Are Considered Insane by Those Who Can’t Hear the Music.


#31 fajw

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Posted 09 January 2013 - 01:20 PM

I'm interested in the new Sound Blaster ZxR.

#32 Nich...

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Posted 24 January 2013 - 12:11 PM

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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#33 .:Cyb3rGlitch:.

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Posted 24 January 2013 - 01:37 PM

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.

"We are a way for the cosmos to know itself." - Carl Sagan
"I do not fear death. I had been dead for billions and billions of years before I was born, and had not suffered the slightest inconvenience from it." - Mark Twain
 
An open mind is willing to consider new ideas, while provisionally accepting those backed by empirical evidence, and provisionally rejecting those without.


#34 @~thehung

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Posted 24 January 2013 - 06:35 PM

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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#35 mudjimba

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Posted 05 February 2013 - 09:09 PM

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, 05 February 2013 - 09:12 PM.


#36 SledgY

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Posted 08 February 2013 - 02:25 PM

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, 08 February 2013 - 02:26 PM.

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#37 @~thehung

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Posted 08 February 2013 - 06:57 PM

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, 08 February 2013 - 06:58 PM.

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#38 SledgY

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Posted 11 February 2013 - 01:33 PM

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.o...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, 11 February 2013 - 01:53 PM.

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#39 Nich...

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Posted 11 February 2013 - 09:41 PM

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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#40 SledgY

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Posted 12 February 2013 - 11:43 AM

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.o...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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