The single worst thing that harms audio quality is excessive compression. It's ruining everything, I'd say. Heck, I'm sure everyone who knows their worth would agree that compression harms audio quality.<p>Recommended reading: <a href="https://en.wikipedia.org/wiki/Loudness_war" rel="nofollow">https://en.wikipedia.org/wiki/Loudness_war</a><p>edit: compression as in dynamic range compression, not data compression like mp3 in audio
I can't hear a difference between 96khz/44khz in it's raw form. However, I can tell the difference from effects in audio mixing. The extra detail can really make a difference in how well an audio effect VST works.<p>I have a 96khz/24bit interface that I use and ATH-M30X headphones, and I can tell a difference between at least some 24bit FLAC files and 16bit highest-quality-possible MP3s. I was mixing my own music and the difference was quite obvious to me. The notable thing was that drum cymbals seemed to have a bit less sizzle and such.<p>Now that being said, if I hadn't heard the song a million times in it's lossless form from trying to mix it, I probably wouldn't have noticed, and even then it didn't actually affect my "experience".<p>I'm one of those guys that downloads vinyl rips as well, but I do that mostly just to experience the alternative mastering, not that I think it's higher quality or anything. (though I have heard a terrible loudness-war CD master that sounded great on vinyl with a different master)
This article hits close to home: before I became a programmer I worked as an audio engineer at a fledgling studio in my hometown.<p>The amount of misinformation / junk-science in the audio world is preposterous. There's a religious-cult of an industry that feeds off the ignorance and placebos of its participants. I have many friends who swear by their What.cd 24/192 FLAC vinyl rips and spend hundreds of dollars on audiophile AC wall outlets. Not to say that there are no differences in high-end audio equipment, but so much of what's "good" is subjective.
First, let me state that I believe that CD audio, played through a modern DAC and quality stereo equipment is pretty much the pinnacle of home audio listening. That is to say, I think 44.1kHz 16-bit PCM audio is plenty good and I'm in no rush to replace my CD collection, nor do I think significant investment in higher bandwidth audio (for playback, mixing and mastering are another story) buys you much.<p>That said, there's one thing the article does not address and that is "beating", or really inter-modulation distortion from instrumental overtones.<p>Instruments are not limited to 20-20kHz. They can have overtones well above this range. Additionally, note that short pulse-width signals, i.e. transients, like drum strikes, especially involving wooden percussion, can have infinite bandwidth. (Not really infinite, but pulse-width is inversely proportional to bandwidth.<p>In a real listening environment (i.e. live performance) these overtones have a chance to interact with one another in the air. It is possible that these overtones may beat with one another and cause inter-modulation products in the audible range. For an example of this, play a 1000 Hz tone through your left speaker, and a 1001 Hz through your right speaker. You will hear a distinct 1 Hz "beat". The audibility of these are largely dependent on listening position and amplitude, but it is possible to occur with instruments. Since most recordings are done using a "close mic" technique (placing the microphone very close to the source) the interactions such as this are never recorded.<p>However, if full bandwidth of the producing instruments is preserved, these interactions of the overtones can be reproduced in a playback environment given equipment having a wide enough bandwidth and degree of quality.
It is not just headphones that are the problem, it is the speakers.<p>People today are often amazed when they listen to CD or turntable content through 70's era crossover speakers. Back in the 70's you'd have a stereo with 2 "speakers" that each had 3 subspeakers for a total of six speakers. The fad today is to have 5.1 sound with a single driver in each satellite, also a total of six speakers. The spatial resolution increase is good for movies, games and TV but surround sound in music is marginal. An amazing number of old "classic rock" recordings were done in quad and anything by Donald Fagan will sound pretty good w/ Dolby Pro Logic, there are some more recent Bjork recordings, but almost everything is mixed for stereo and what you loose in frequency response is not compensated by anything, except perhaps the ability to produce more volume with more speakers.
If you want to know more, Monty made one of the best intros to digital sampling I've ever seen: <a href="https://www.xiph.org/video/vid2.shtml" rel="nofollow">https://www.xiph.org/video/vid2.shtml</a>
I have a pair of Roger Sound Labs studio monitors for my speakers at home. I got to look at their insides when a technician was replacing a blown midrange speaker (they have a "lifetime" warranty, however that warranty expired when RSL did). Looking at the cross over filter network I could see a network selecting for frequencies > 20khz and it was shunted to a resistor. I asked about it, and the reponse was exactly like the authors, by filtering out signals higher than the tweeter could reproduce, they improved the listening experience.<p>It made sense to me, and I love how the speakers sound. Understanding is not inserting distortion makes even more sense.
Why do high quality DACs clearly sound better then? And they sound better with better files. Maybe it really is all in my head but I mean listening to a £20000 hifi the other day (vinyl) really just shocked me.<p>I was listening to Marvin Gaye on my friends system and I could hear that there were several different backing singers all moving and at different distances from the microphone.<p>Are there any double blind trials anywhere of Vinyl/CD/24-192khz with super high end hifi systems? Mostly I see people suggesting that these tests are performed from the phono output of a mac with a pair of average ear buds...
From my experience, what matters more than sample rate is 24 bit vs. 16 bit sampling in the recording/production process. Using heavy compression and EQ can mean that very quiet sounds can become louder, in this case 24 bit recording is ideal. Sample rate wise, anything above 40khz is fine for most ears (I've probably lost a few khz in the upper range anyways) Another note is that most converters operate at a multiple of 48K, so it makes sense to use 48/96khz if you are recording. It all comes down to how much disk space you have, and want to use up.
<p><pre><code> Because digital filters have few of the practical
limitations of an analog filter, we can complete the
anti-aliasing process with greater efficiency and
precision digitally. The very high rate raw digital
signal passes through a digital anti-aliasing filter,
which has no trouble fitting a transition band into a
tight space.
</code></pre>
I always thought <i>digital</i> anti-aliasing filters were
creatures from a fairy-tale world. Much talked about
but no one has ever seen one.<p>My understanding: If you have a an analog filter of a given steepness the only way to further reduce aliasing effects digitally is oversampling. Or less steep (cheaper) analog filter plus oversampling is the same as steeper (more) expensive analog filter. People tend to say <i>digital</i> anti-aliasing filters when they really mean oversampling.<p>"24/192 music downloads make no sense" seems to be a thoroughly researched and carefully written article. It explains oversampling very well, possible confusion with digital filtering (anti-aliasing or not) is out of question. But then it goes on to talk about <i>digital</i> anti-aliasing filters, which makes me afraid I could be wrong.<p>Do <i>digital</i> anti-aliasing filters exist?
Try what age is your ears
<a href="https://www.youtube.com/watch?v=VxcbppCX6Rk" rel="nofollow">https://www.youtube.com/watch?v=VxcbppCX6Rk</a><p>Or generate a tone sweep in audacity.
Generate->chirp
<a href="http://www.audacityteam.org/" rel="nofollow">http://www.audacityteam.org/</a><p>You loose the ability to hear high frequency sounds as you age.<p>Personally I can hear up to about 14kHZ
Wouldn't this question be answered with a large-scale double blind trial?<p>If more people prefer the sound at the higher bitrate and sampling rate, then that's the better format, even if there's no technical reason why that format is superior.<p>Much like how some people prefer the "warm" sound of tube amps, even if that means more distortion.
I can hear insects and buzzing electronic devices, and my partner thinks I'm crazy some times. Thinking I might have golden ears I tested * my range and I could hear up to 18kHz.<p>* <a href="http://onlinetonegenerator.com/hearingtest.html" rel="nofollow">http://onlinetonegenerator.com/hearingtest.html</a>
Some [consumer] digital low-pass filter can benefit from higher sampling rates, leading to an overall better representation of the analog signal up to 20kHz. But there are diminishing returns as the filter "folds" the octaves above 22kHz; A rate of 96k for certain lowpass filters is better than 48k, but at some point there's little (if any) benefit by going to 192k or 384k. For recording studios, go as high as you can in both bit-rate and bit-depth. Especially when you're processing the signal "in the box". Give the software as much data as possible to operate without introducing errors and artifacts. There are diminishing returns there as well, but RTFM for (for example) UA gear and software and you're good to go.
24/192 lossless is a digital Veblen good; some people will pay more for it (and/or the HW to play it & store it), and almost all of them will enjoy it more, if only because it costs more. Whether it actually sounds better is somewhat tangential.
I just recently purchased Izotope Ozone 7 advanced. One feature it has is "codec preview" which lets you "solo" the codec artifacts for MP3 and AAC format. Even at high but rates it's amazing how swishy bit reduction sounds. It also made me realize what I was hearing with mp3s was artifacts from compression. That said, it's not unlike tape hiss or vinyl noise. In fact I think it can have its own charm and in some cases make the music sound more full. It's also probably why 24/192 digital audio can sound so "cold" or lifeless.
I love the idea the author mentions in passing of a dedicated speaker assembly for ultrasonics. This seems like something that could be a huge margin business, and the parts costs would be as low as you wanted.
They <i>are</i> useful if you're resampling them or editing them, but I doubt that's something consumer music services are overly concerned with.
There's a big difference in impulse response with different sample rates, any one can see it on a oscilloscope, I bet some one can hear the difference.<p>Those who don't have a oscilloscope can see the picture here:
<a href="http://i.imgur.com/wY0wzcW.png" rel="nofollow">http://i.imgur.com/wY0wzcW.png</a>
My third time reading this, and a new question popped into my head: Are there any volume adjustments (on software or hardware) that take into account the pain threshold curves? That is, volume adjustments that aren't flat, but that will attenuate the frequencies that will cause discomfort at the lowest volumes?
So, no point in 24/192 because it makes no difference in playback... but having lossless downloads is important in part for enabling remix culture? There's a bit of a double-standard here. Maybe I can't hear 24/192 audio, but isn't it better input for sampling?
To what can I attribute the consistently horrible quality of 64kHz streams ten or fifteen years ago? Would that fall under the "bad encoder" bucket?<p>Edit: christ, I mixed up bitrates (e.g. 192kbps) with sampling frequency (e.g. 192kHz) again. I was referring to 64kbps streams.
Everything else is fine and good in the article, but I <i>can</i> see the infrared in the Apple remote (and all the other IR remotes I've tried). It's faint, but plainly visible. Am I the only one?
This article really misses the facts of the Nyquest-Shannon theory.<p>In order to decimate a signal to 44.1 or 48khz, and preserve high-frequency content, high frequencies need to be phase-shifted.<p>This phase-shift is similar to how lossy codecs work.<p>For what it's worth: I'm a big fan of music in surround, and most of it comes in high sampling rates. When I investigated ripping my DVD-As and Blurays, I found that they never have music over 20khz. It's all filtered out. However, downsampling to 44.1 or 48khz isn't "lossless" because of the phase shift needed due to the Nyquist-Shannon theory.<p>I still rip my DVD-As at 48khz, though. There isn't a good lossless codec that can preserve phase at high frequencies, yet approach the bitrate of 12/48 flac.