MP3 'died' and nobody noticed: Key patents expire on golden oldie tech Key patents held on the MP3 audio format expired last month – and it's taken three weeks for anyone to notice. And the Fraunhofer Institute has declined to renew the intellectual property it owns on the MPEG Audio Layer III technology – as well as terminating its licensing programme. So yup, now you can use MP3 encoding in …
"If you're looking for bleeding-edge music formats today, have a look at MQA, or Master Quality Authenticated (MQA)"
Audiofools please form an orderly queue here where you can talk complete bollox.
PS don't forget to stick used egg boxes all over your walls to kill unwanted standing waves as a complement to the ridiculously expensive oxygen free copper cables you bought.
Stereophile, a synonym for gobbledegook talking wanker.
When you're streaming music, don't use WiFi, use a $10000 gold Ethernet cable made for audio fidelity. Anything else could play havoc with your lossy compression algorithm.
" don't use WiFi, use a $10000 gold Ethernet cable made for audio fidelity"
Ethernet! What are you smoking? It's common knowledge that Ethernet cables use a linear signal and so induce audible negative feedback at primary and secondary resonant frequencies!
Any proper audiophile knows that you need a loop topology like Token Ring for a digital music network, and with Token Ring you get the added benefit of being able to tune the token to your prefered frequency to create a more ambient tone (as long as platinum carbide termination is used so that the token doesn't fall out of course.)
<Coughs>
"Through all the bullshit people fail to actually see what matters, a good tune."
A good tune, which is subjective, yes.
Then followed by good production, that helps.
Then good production is rewarded by good playback.
You can skimp on any of that. But then it limits what you can achieve. THis is why 24bit and 96khz exists. If not higher.
Snake oil salesmen, yes. Fine. But you may as well get upset with people for liking Bieber, as paying a lot of money onto something that they *think* makes it sound better. If they think it does, that is all that matters in that context. That is the way subjective appreciation works. It is the way it always works. If you are always worried about the science behind listening to music, then you too are repeating the same pattern as those that worry their copper interconnects are may have too much signal degredation over the 500mm and want to replace them with Orgonite connectors.
>But then it limits what you can achieve. THis is why 24bit and 96khz exists. If not higher.
I see you joined the queue, you may want to look how speakers and human hearing works as to why this is irrelevant.
I challenge you £5,000 to tell the difference between 320kbps MP3, CD, 16 bit FLAC and 24bit 96khz in a blind hearing test conducted by me or Mr James Randi.
Pi is now quoted to 2.2 x 10^13 decimal places but that doesn't make it any better at landing you on the Moon.
MASTERING.
I know I can't tell the difference, but signal degrades every time you do anything with it, therefore you MASTER @ 24bit 96khz to ensure you don't lose any of it.
If you do not understand this concept, then you don't really stand from any position to lecture anyone on audio.
EDIT: I didn't specifically mention mastering before, I now realise, but I was talking from the standpoint of production. This is why 24bit and 96khz exist. If you tried to do this at any less, you would be working with a degraded signal and by the time you finished, the end result would be poorer. Think of it like working with a dirty lens. You wouldn't notice this in the picture, but if you continued taking a picture of the picture over X amount of times, you would see the difference the dirty lens made compared to a pristine lense.
>I know I can't tell the difference, but signal degrades every time you do anything with it, therefore you MASTER @ 24bit 96khz to ensure you don't lose any of it.
Quantum physics, it's always degraded no matter what you do and will always be an approximation. Heck why not use 256bit and 1Ghz sampling ?
If only I could record the position and energy of every particle at every moment in the room then I'd have the perfect audio recording, oh wait hang on......cue Professor Heisenberg.
"therefore you MASTER @ 24bit 96khz to ensure you don't lose any of it."
Ehm, no.
Mastering has nothing to do with audio degradation but more so with trying to enhance the audio signal. For example by 'pushing' lower audio frequencies when certain bass instruments are used (bass drum, bass guitar, cello, etc.). Another important aspect of mastering can be to try and ensure that certain higher signals don't "mix" or get canceled out. Sometimes that's done by pushing higher frequencies when lower regions dominate (think about your bass drum which can block higher frequencies).
Also: you optimally master material which sits around -6dB. Bit rate and/or frequency is pretty much irrelevant.
Currently it's either/or: either XXXbits at 96/48/44.1KHz or YYYMHz with 1bit:
https://en.wikipedia.org/wiki/Direct_Stream_Digital
"A further extension to the development of DSD is DSD512, with a sample rate of 22.5792 MHz (512 times that of CD), or alternatively 24.576 MHz (512 times 48 kHz)."
There used to be RHCP - Blood Sugar Sex Magik disk's image in DSD128 format IIRC on torrents, you might want to check it out (searching for a player capable of playing this would be fun, they said).
Forget all this boll*x about quantum physics and sampling rates.
Wot you need is VALVES, and OUTPUT TRANSFORMERS!
There's nothing like the non linearity of a BH loop and a tranny running out of iron to adequately compensate for the inherent non linearity of the human ear canal.
The crossover point of a pair of kt66's does need to be adequately matched to the unique characteristics of the individual listener’s middle ears though - which is why the old lags always insisted on TWO valves, natch.
And, of course, the warming of the electrons adds appreciably to the improbability of any distortion, which the audiophiles of old determined by purely empirical methods, predating by several decades the elegant mathematical proof of infinite improbability.
And, of an additional course, while you might well refuse to by that LP because it was scratched, it'd still play, the additional rhythmic click only adding to the enjoyment of the ambience provided by the dust on the record.
And sod all this rubbish about oxygen free copper - How do you expect the bloody electrons to breath without oxygen - all you'll get is a thin wheezy sound.
Anyways it's not a to do with the conductors - it's the insulation. You need rubber, not that nasty plasticy stuff most folks will try to fob you off with these days.
NURSE!
"Wot you need is VALVES, and OUTPUT TRANSFORMERS!Did you ever listen to a 1960s valve amp and compare it to a 1970s transistor amp? Valve harmonic distortion is even harmonics, while transistors generate odd harmonics that are far more noticeable and annoying.There's nothing like the non linearity of a BH loop and a tranny running out of iron to adequately compensate for the inherent non linearity of the human ear canal. "
Even today after decades of refinement amplifiers are far more variable than I thought they might be. My Rotel receiver's fancy electronic volume control decided to die after 16 years of service and it's non-replaceable. Being short of readies until the farm sells, I replaced it with a more recent 2nd hand Sony. I'm not a golden-ears by any stretch of the imagination, but it's best described as very ordinary.
As for oxygen-free copper, surely elemental copper is oxygen-free by definition. Copper combined with oxygen is copper oxide. The reason "monster" cable makes a difference is down to Ohm's Law. Bass reproduction requires high currents and the lower the resistance of the conductors, the less lossy the transmitted signal. And it is audible even when you're not a golden-ears.
In 1969 Quad (as they were then) were testing their new current dumping amplifier. The did double blind tests with golden eared journos from the HiFi mags of the day.
They tests their MkII valve ampr (1% distortion) , their 33/303 transistor amps (0.1% distortion) against the 404/44 (0.01%)setup.
Not one of the golden ears could accurately tell the difference in double blind testing. The golden ears response was to refuse to do double blind testing and put themselves out of the valuable work of bullshitting about HiFi.
"They tests their MkII valve ampr (1% distortion) , their 33/303 transistor amps (0.1% distortion) against the 404/44 (0.01%)setup.Not one of the golden ears could accurately tell the difference in double blind testing."
It would be interesting to know what music they were listening to. There was a test of this conducted in the USA back in the early 1970s with a doctored Phase Linear amp (IIRC). The listening panel was presented with various music recordings — flute, piano, violin, voice etc — at different levels of harmonic distortion (THD). THD had to be quite high ~10% to be audible on some content, but was clearly audible at 1% on different content.
It's worth noting that THD varies with the output power of the amplifier. Traditionally it was measured at the amplifier's maximum rated output. Class B amplifiers generate high distortion at quite low levels. Most of the time amplifiers are idling along at a fraction of their rated output.
Two other distortions had already become apparent back then. Crossover distortion in the common Class B amps of the day, and transient intermodulation distortion. The first was overcome by Class A designs though these were only half as efficient as Class B and the second by making amplifiers capable of handling frequencies well in excess of the standard 20 kHz which being beyond the limit of human hearing was considered to be more than adequate.
Damn straight. I made sure to get the headphones that are perspiration-resistant for exactly that reason. Normally the wires' plastic would gradually mop up the skin oil off your neck and sideburns and become brittle over time; but the signal degradation is really your first clue. That these will remain supple for a while longer is just the cherry on top. Someday, someone will get wise and use copper silver deposition to assemble a conductor strand inside a fibre optic line, making that into the sheath... then there shall be eargasms \o/
"but signal degrades every time you do anything with it"
No it generally doesn't in the digital sampling rates you are referring to - a digital copy of a digital copy is identical...And you would normally be using digital processing on it.
In general imo 96KHz sampling rate is a waste of money. There is no extra audible audio information captured in 96KHz sampling - and the more common 48Khz is enough to allow a reasonably gentle filter roll off from 20KHz when going back to analogue...
However, 96KHZ usually uses a 24 bit sample size that when compared to 16 bit is advantageous as it allows a greater dynamic range...
<draws breath>
Right. Digital audio is described with two sets of numbers:
The SAMPLE RATE, which is a measure of the number of times an analogue wave form is measured per second, i.e frequency. This is your 44.1, 48, 96 KHz number.
The BIT DEPTH. (NOT, repeat NOT 'Bit Rate'.) This is your 16 bit or 24 bit number, and details the amplitude of the analogue waveform at the point where it was sampled. A 16 bit sample has 65,536 possible values (2^16), a 24 bit sample has 16,777,216 (2^24) possible values.
Because Bit Depth is a measure of a wave's amplitude, it can be used as an indicator of a sound's loudness, and this is where using 24 bit makes sense. All audio systems have two boundaries: the lower being the point where a signal becomes indistinguishable from background hiss (the noise floor) and the upper, reached when the signal level becomes so high that the system becomes unable to process it, causing distortion ('clipping', as the peaks and troughs of the waveform are clipped.)
A greater bit depth offers a wider Dynamic Range (more 'headroom') when digitally recording a sound - less background hiss (i.e. a lower noise floor) means that you can record extremely quiet sounds clearly; more numbers (2^24 at 24 bit compared to 2^16 at 16 bit) means you can record much louder sounds without the signal clipping.
In the olden days, even the best studio equipment was noisy, meaning that there was much less headroom available when recording. Studio engineers spent a lot of time managing input levels so that a recording would cleanly capture a performance without the quietest parts being lost in background noise or the loudest parts being clipped. 24 bit recording means that studio engineers don't have to do that any more, as the format offers them enough room to cleanly capture a performance, and if a guitarist decides to turn an amp up to 11, the engineer doesn't particularly have to worry about the signal clipping.
If not more importantly, considering that the vast majority of music made today is created almost entirely through software (including those beloved classic album remasters), the extra headroom that 24 bit offers means that a producer is able to pile on the signal processing effects without being forced to degrade the signal quality of the piece to do so.
24 bit audio is now indispensible in audio production, and people are already moving up to 32 bit floating point workflows.
What about for audio playback? As we know, 24 bit gives us a wider dynamic range - the difference between the quietest sound we can detect and the loudest we can cleanly process. Say you had a 24 bit audio file that consisted of a tone just distinguishable above the noise floor, which then increased to the point where it clips, and that you played this through a pair of capable speakers. By the time the playback finished, you would be in agony, and your hearing would be permanently damaged.
No piece of audio for playback would ever be put out that uses the entire dynamic range 24 bit offers. It would permanently damage customers' audio equipment and their hearing. All a 24 bit file gives you that is missing from a 16 bit recording is masses of dead, empty headroom in which there is nothing to hear, and that will never, ever get used.
Anyone who tells you that listening to a track rendered as a 24 bit file is 'better' than listening to the same audio rendered as 16 bit is a liar or a fool, and either selling you something or trying to justify something expensive they've bought.
""but signal degrades every time you do anything with it"Here's an experiment you can try at home. Convert an MP3 to WAV, then recompress to MP3. Rinse and repeat several times. The signal deteriorates at every step as becomes apparent the more often you rinse and repeat.No it generally doesn't in the digital sampling rates you are referring to - a digital copy of a digital copy is identical...And you would normally be using digital processing on it."
Because my dear deluded friend, MP3 is a lossy format. It throws information away every time it encodes. So your loop experiment just shows you know nothing about digital formats. Do your files on your computer 'degrade' when you copy them? No? Try to figure out why.
"Because my dear deluded friend, MP3 is a lossy format. It throws information away every time it encodes."Precisely, hence my comment that digital processing does degrade the signal, contra the original comment that digital processing does not degrade the signal. It's not me that's deluded.
Also worth bearing in mind that it doesn't take more than two iterations to tell the difference, so no, MP3 is nowhere near as good as FLAC.
In theory, given a miraculously good decoder and very specific fixed rate encoding settings matching the original encoding, MP3->lossless->MP3 could actually be lossless. The original encoding would have removed the features MP3 removes.
When it looked for masking, those frequencies wouldn't be there to be removed. When it looked for bins to eliminate it would find there were just enough present it wouldn't need to eliminate any of them. The bins would match those after 1st encoder decimated them and generate the same quantisation factors.
...given a miraculously good decoder. In real life you'd see some quantisation errors slowly accumulate. In real life the settings wouldn't really match.
"In theory, given a miraculously good decoder and very specific fixed rate encoding settings matching the original encoding, MP3->lossless->MP3 could actually be lossless."The original experiment was done with Steinberg Clean (Fraunhofer codec). I have no idea where the artefacts arose, in the software or the DAC firmware, but they were definitely there. Much to the chagrin of the dude who first told me "digital processing does not degrade the signal".
Reminds me of when the Sugden A21 amplifier was theoretically inferior to its rivals because it had an order of magnitude higher (or more) total harmonic distortion. In the real world it was the best of breed.
The point of the high bit-count high-sample rate audio in the studio is that the signal gets filtered, scaled, and maybe even sped up/slowed down before outputting to CD.
Just like calculating with rounded-off numbers, starting with 16-bit 44kHz data would lead to rounding errors in the output, which increases quantisation noise. The Nyquist filter for 96kHz sampling can be a fairly simple affair, as it doesn't need to roll off particularly quickly. The recovery filter in CD players needs to be quite steep to prevent >22KHz sidebands in the output, which, whilst being inaudible themselves, can cause intermodulation distortion in less-that-perfectly linear devices downstream. Like speakers, for example. As a result, the phase behaviour can be a bit squirrely.
> m0rt: but signal degrades every time you do anything with it, therefore you MASTER @ 24bit 96khz to ensure you don't lose any of it.
Not so. Digital is digital - you may reclock it, but there should be no bit-lossage.
Also, it's worth remembering that most studios have analogue front-ends, i.e. they are analogue through the mixing desk up to the final ADCs that convert the mixed and engineered recording to digital.
During that process, the ANALOG signal will travel from the microphones (analog) through maybe 100s of opamps (all analog - probably NE5534s or some of the more exotic Burr-Brown/TI/Linear Tech/Analog Devices ones), though a mixing deck (probably analog) where someone with expensive (analog) ears and (analog) monitors will mix it into something the producer likes, before committing it to digital.
So, the argument is about turning that hugely-messed-around-with digital representation back into something analog that you ears can handle.
I've spent many years designing amps, solid-state (linear & class D), valve and hybrid as well as a lot of speakers. At the end of the day, a technically perfect amplifier is a straight piece of wire with gain - it is possible (fairly easily) to design such an amplifier. Doug Self, one of the real gurus of professional audio design and a terrific engineer, hates audio-phoolishness and some years ago published a design where distortion levels were vanishingly small - these are known as "Blameless Amplifiers" (look them up). But they sound cold, because oddly, people like distortion of various types, specifically with speakers (which are a HUGE source of distortion) - this is also the reason people like the sound of valve amps - they sound "warm" due mainly to 2nd harmonic distortion.
The human brain & ears are infinitely complex & subtle - this is why audiology is such a complex subject. People go to enormous trouble and expense to get the "perfect" sound, but it's mainly b*llocks - it might be right for them, but it' all in the mind and won't be right for someone else. If you spend $5,000 on speaker cables, your mind will probably tell you they make a huge difference and sound fantastic; you listen to someone else's $5,000 speaker cables, and you'll probably not hear any difference.
People live to heap scorn on gold plate on connectors, but the idea is that it is tarnish/oxidisation resistant for contact connections. Gold plate is being used plenty inside electronic devices where you can't see it for the same reason. It has nothing to do with bling or signal. Edge connectors on 1980s computers were gold plated.
However, it should not make cables expensive because it's a few microns and very cheap.
"Not so. Digital is digital - you may reclock it, but there should be no bit-lossage."
Odd that the industry is so obsessed with error correction then, don't you find? Bit loss is common, very common. Error correction means that ultimately that doesn't matter because the copy will usually reproduce the original perfectly despite the loss of bits. The copy on disk may or may not be a bit for bit representation of the original data stream but the information should be.
> therefore you MASTER @ 24bit 96khz to ensure you don't lose any of it.
No you really don't. You might choose to record at 24bit/ 96, but CDs are 16/44, so you master to that.
85dB has been the maximum level allowed for headphone levels on consumer music equipment for a few years now, well within the 96db of 16bit.
You cannot lose what isn't there in the first place. 24bits is 144db of dynamic range, so unless you are in the business of recording Saturn 5 take offs, that's not the reason, even then, finding a sound system that could actually produce that dynamic range would be a challenge. Your ears tend to fail with permanent effects above 120dB.
SACD potentially allowed 105db range and higher frequency range, but people really couldn't tell the difference and a year long AES study showed that in double blind tests, the chance of correctly identifying CD vs SACD was barely 50%, no better than guessing.
In the early days of digital recording, that 44.1K sample rate lead to some fairly unpleasant filter designs which lead to poor reproduction, but it's perfectly capable of recording sounds that beyond what most adults can perceive.
The irony is that the loudness wars, caused by marketing driven desires to be the loudest, have actually reduced the dynamic range of music today. Mastering engineers like Bob Katz have tried fighting back, but with limited success.
Here let me google that for you:
"As a result, the European Committee for Electrotechnical Standardisation (CENELEC) amended its safety standard for personal music players.
Now all personal music players sold in the EU after February 2013 are expected to have a default sound limit of 85dB. "
http://www.bbc.co.uk/news/health-21294537
"Here let me google that for you:"Here let me google that for you:
"The decibel ( dB) is used to measure sound level, but it is also widely used in electronics, signals and communication. The dB is a logarithmic way of describing a ratio. The ratio may be power, sound pressure, voltage or intensity or several other things. Later on we relate dB to the phon and the sone (related to loudness). But first, to get a taste for logarithmic expressions, let's look at some numbers."What is a decibel?
*sigh*
"... to ensure you don't lose any of it."
you are still aliasing anything between any two bit counts.. step from 14 to 15, what about 14.3? it cannot be _faithfully_ encoded in a digital form .. . you have lost immediately by sampling. In a lab, virtually any sampling resolution could be shown to be different from the original. I get what you are saying but you need to qualify "perceived loss"
If audiophile reckon they can tell the difference if the signal comes through cable with oxygen in the insulator then this matters no?
Careful there - no-one has ever been able to tell the difference between un-encoded* 16 bit or higher but MP3 adds features to the stream that are audible with experience and people can tell the difference in double blind tests.
* presumable flac would be undetectable too but that assumes the decoding is carried out seamlessly and the spec doesnt cover how the operating system works so it is possible for flac decoding to add very small gaps which may be unnoticeable by some but observable by others.
Well, maybe people do have the right to waste their own money on placebo bullshit, but that still doesn't make it anything other than placebo bullshit, and that bears repeating loud and often so their bullshit doesn't mutate into a cultural norm.
The argument against audiophile gibberish is basically the same as the one against religion. It's not about what they believe, it's about what they tell everyone else to believe.
"So when you get all stressy about the lack of protection from being exploited in this way, am I still to fuck off?"Frankly I don't get all stressy when I'm being ripped off purchasing "cheap Chinese shit on ebay". I'd rather pay $AU10 for two camera batteries than $AU140 (plus a stocking fee) each from Nikon Australia. I know who I need protecting from; you obviously don't.
Never mind the wheels on my goalpost (whatever that means), how big is your ego that you know more about my needs than I do?
The Register 
Biting the hand that feeds IT
Copyright. All rights reserved © 1998–2024
