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Lo-Fi Phones and the Future
bossanovalithium writes "Back in 1936 — 74 years ago — boffins accepted that about 3.3Khz was the accepted frequency that telephone calls are going to run on and it's been like that, generally, ever since. Call quality is reasonable but leaves a lot to be desired. Think calls from Skype to Skype where quality is often crystal clear." It's crazy to me that (for people with decent mics at least) Ventrillo sounds better than corporate conference calls.
Who needs be My post would be MORE insightful, but the the slashdot effect prevents me from reading the article, and the slashdot code of ethics requires me not to.
Patience is a virtue, but haste is my life.
I live in 3rd world country and our major cellphone networks support hd-voice codecs.
boffins?
Do everyday Brits actually use this word conversationally?
Looks like their server has Lo-Fi bandwidth....
Otherwise known as... Slashdotted. I hope their ISP doesn't put the hammer down on them.
ASCII tastes bad dude.
Binary it is then.
Slashdotted already?
hmm, well it is running IIS...
No comments but the link has already crashed from getting slashdotted?
Just click here and avoid the Slashdotting...
coding is life
Sorry Taco, I disagree - I've yet to be told during a conference calls to go handle "many whelps."
I can't even read the referenced article but I can tell you the phase ""Back in 1936 — 74 years ago — boffins accepted that about 3.3Khz was the accepted frequency that telephone calls are going to run on" is totally wrong.
What they meant to say was that the relevant bandwidth for understanding speech would be from 100Hz to 3.4kHz. Making the required bandwidth be 3.3Khz.
What's crazy to me is the summary goes from 3.3khz to Skype in the quoted portion and somehow jumps to Ventrillo with nary a through of a segue.
Sig Battery depleted. Reverting to safe mode.
the only voip solution that withstood long (more than 8 h) sessions and has a very good audio quality over that time in my experience is the mumble project
My experience with Skype, VOIP, and even to a lesser degree cell phones is that they all have latency worse than landlines. Is this actually true?
We were considering switching our business phone lines over to Time Warner voip. I talked to one of their people on the phone. My side was landline, theirs was time warner voip. The delay was awful. We kept talking over each other. If that's the best Time Warner can do, I was very not impressed, and as a result was still have our more expensive landlines.
Is there anything to my complaint, or have I just had bad luck??
So how many boffins died to bring 3.3Khz to our phones?
"I use a Mac because I'm just better than you are."
This has got to be up there in the competition. Doesn't layout a summary of the article. Offers an opinion about some piece of software I've never heard of. No hint of whether or not there's a proposed solution.
Bizarre.
"Who is the Journal of Quantum Physics going to believe?" --Stephen Hawking
At 3KHz, with compression, you can now record every conversation, from birth to death, of a connection. Think about who wants that data. I would guess that from the moment you aquire your first cell phone contract, the providers are saving all your conversations. What's the point of a wire tap when that data is available upon request? In our post 9/11 world, I would be amazed if it doesn't already work that way.
A conference call over Vent would be funny.
+Guild Chat
+Rankor's Room
+Raid1
+Raid2
+Raid3
+Business Conference
"Now if all the raid members would kindly leave out channel so we can get down to business... No Stan, get out of Raid1 chat..."
For some reason I have a problem with low quality audio such as AM and telephones. While other people have no problem hearing what's said all I hear is something that's recognizable as a human voice, but doesn't seem to be saying anything comprehensible. I've had my hearing tested and it's actually above average. So what gives? My best guess is that I grew up in a hi-fi world where FM quality was the bottom end and "mono" didn't exist. My ears are trained for frequency and dynamic ranges well beyond what you get from an ancient telephone with tiny tinny little speaker or a mono AM radio. So when confronted with audio like that it perceptually falls into the "noise" range.
Things are better if I can use a decent headset with a phone, and all problems go away when I can use a high-quality audio service like Skype. Have any studies been done on people that did not grow up with the low audio quality of telephones and AM radio? Is my guess close to the mark? Is it just me that has trouble understanding lo-fi audio or is this common to people who grew up not having to listen to lo-fi audio?
Whatever my root problem is, I know I'm not the only person that could benefit from improved audio quality in telephone calls. Get to upgrading those systems, phone companies. It's decades overdue.
But in actual practice, if you have a $40 Wireless N router, an iPad makes a very cheap phone.
And it comes with the ability in the new model releasing later this year to use iFace to share pics while you talk with iSkype.
Computers were originally used mostly for accounting, calculating missile trajectories, and for other stuff, but we don't use them to do that now, for the most part.
-- Tigger warning: This post may contain tiggers! --
Don't do it. The FCC changed the way television works, and look what we have now... none of my old TVs work anymore! I dread the day when my 1936 Western Electric 202 desk set stops working just because some kid wanted to listen to his girlfriend yammer in Hi-Fi.
The determined Real Programmer can write Fortran programs in any language.
Not bloody likely. Maybe in a perfect world with computers directly connected but every real world example of Skype that I have seen was awful.
the problem has been solved yet not been implemented widely. It's called ENUM and freely available and open. No need for proprietary XConnect stuff to implement this functionality, it's based off DNS and thus already has a widely available penetration. All people (and large corporations) need to do is actually use it.
Custom electronics and digital signage for your business: www.evcircuits.com
Back in 1936 — 74 years ago — boffins accepted that about 3.3Khz was the accepted frequency that telephone calls are going to run on and it's been like that, generally, ever since.
Back in 1936, nobody expected they would have to scatter from the Lich King's defiles while a single player messing that up would cause a wipe.
http://www.techeye.net.nyud.net/business/how-and-why-telephones-are-going-to-get-a-whole-lot-better
I was pondering this exact stuff just today at work, since a phone call sounded kinda crappy, barely acceptable until I needed to involve 2 more people and put it on speakerphone, it became so bad we had to give up. I dropped the phone call, switched to skype, and damn what a big difference. The crappiness of POTS is ridiculous indeed, and although I see the need for compatibility, it can't die soon enough.
By the way, if you like Ventrilo, try Mumble, which, apart from being free and open source, which can't hurt according to the /. crowd, has really awesome sound quality, and you can setup your own private instance in minutes. Plus, for the MMO crowds, it has extremely low latency, awesome echo echo echo echo cancellation and built-in auto volume normalization (helpful when That Loud Guy Without Headphones keeps pressing his PTT and everyone's in pain)
Vacuum cleaners suck. Kings rule.
Shhh, if Jobs finds-out you know the game-changing "camera" feature of the iPad 2.0gs some poor guy in China's going to get axed.
--- Need web hosting?
While bandwidth is low, that's not the big problem. Quality is really hard to fix over networks with time jitter. Which is why VoIP and cell phone voice quality frequently suck. The best phone audio today is from an ISDN phone to an ISDN phone - end to end uncompressed full duplex digital with hard bit timing synchronization. (ISDN voice never caught on in the US, but it's widely used in some European countries.)
Wire-line telephony is 8 bits sampled at 8KHz, so the highest potential bandwidth is 4KHz. Compare CD audio, 16 bits sampled at 44.1 KHz per channel. Cell phones are worse; they're usually compressed down to 9600 baud or so. There are some high-end video conferencing systems with higher-bandwidth audio, but they're rare.
Sorry Taco, I disagree - I've yet to be told during a conference calls to go handle "many whelps."
If you can't even make it to the whelps, then you don't know how to play, and that's 50 DKP-minus
It's crazy to me that (for people with decent mics at least) Ventrillo sounds better than corporate conference calls.
Woah, maybe better than YOUR corporate conference calls, but definitely not all. Apparently you've never heard of "HD Voice".
i use teamspeak alot and whenever i have to talk on a regular old phone it pains me from the quality... a course im in the middle of no whare and line quality is bad and so forth so it may not be as bad elseware
epic sig..... ya i got nothing
Is this non-native English thing? ~4Khz was (is?) the bandwidth for telephone voice transmission, not "frequency".
Fuck systemd. Fuck Redhat. Fuck Soylent, too. Wait, scratch the last one.
Coral Cache version, AKA http://www.techeye.net.nyud.net/business/how-and-why-telephones-are-going-to-get-a-whole-lot-better.
AC post not to karma-whore. Remember kids, add nyud.net to a server's name when it's dead, and submitters please do the same before the page gets published.
The bare wires to the phone support a lot more than 3.3Khz of bandwidth, so there's no reason why your old phone couldn't continue to work. Theoretically all the work could be done in the telco equipment. Worst case, you'd have a little breakout box with RJ11 one one end and RJ45 on the other.
Hi!
In a year or two, most GSM/W-CDMA networks will be upgraded to WB-AMR codecs.
Orange is already using it in Moldova and London, others are testing.
It is marketed as High Definition Voice.
WB-AMR uses 16 kHz sampling instead of classic 8 kHz . Together with better voice compression,
higher quality of voice is using same capacity (say, 12.2 kbit/sec) as we use today.
Of course, PCM is out.
Both sides of connection must support WB-AMR, and everything in between as well,
so for few years it might not be available across different networks.
If one terminal can not use it any more (maybe due to handover to GSM cell not supporting WB-AMR),
fallback to AMR/EFR is made on both sides, using 64k/56k PCM inbetween.
Technology is avaialble for quite same time, but terminal vendors are slowing it down.
Some 20% of all terminals have to support it, otherwise it makes no sense for operator
to buy all SW needed to implement it network wide.
Funny: good old GSM will soon get higher voice quality as ISDN.
73
After a recent PBX upgrade that automatically enabled G.722, I was asked to disable it because the users perceived the call quality as being worse than G.711 and G.729. What I realized is the users were hearing higher frequencies and could also hear more background noise (think Sprint pin drop commercials) than before and were interpreting it all as a noisy connection. The response to the explanation of the situation was "Turn it off".
I don't have an issue with the frequency range, but certainly do with latency, and the lack of true duplex any more!
I find (found) that talking on a true analog line is MUCH easier than any digital line today - be that Skype, cell phones, or even land lines in most countries. I'm always amazed when traveling abroad when I make a local call on a truly-analog system how much nicer the experience is!
With today's systems in "Westernized" countries, you can't even have an effective 2-way conversation. The duplex performance sucks - you can't hear anything while you're talking. Add to that a small but noticable delay, and you have to resort to long pauses between sentences to ensure you don't talk over one another.
Am I the only one that notices this? It's AWFUL compared to what it was like 20 years ago.
MadCow.
I used to have a sig, but I set it free and it never came back.
"Call quality is reasonable but leaves a lot to be desired."
If you think call quality is inadequate on Plain Old Telephone Service, have you ever tried using wireless phones? On POTS we used to commercials that promised "you can hear a pin drop". Now it's "can you hear me now?"
http://alternatives.rzero.com/
Even in the dawn of telephony, frequency response was a significant issue. Besides the poor quality of transducers, the lines themselves weren't very good. Twisted pairs would have been nice, but early telephone wasn't twisted to improve common-mode rejection, it was twisted to keep the pairs together. Common residential service used something approaching zip cord from about 1960 on, maybe earlier. This isn't even twisted. You wonder why your DSL service is so crappy? I wonder how it even works at all. 10Base-T would barf on 30 feet of straight-line zip cord, and there is a good chance your house has 60-80 feet of it from the pole to the NT1. My first ISDN service at home was a fiasco, with load coils and conditioners being ripped out and new cable strung from the street to the complex demarc.
Frequency response is not the same thing as bandwidth (though they are directly related), but for telephone a 300-3300Hz response is intelligible and manageable. Doubling it to 6500Hz doesn't do a whole lot except consume bandwidth and marginally improve intelligibility. If you want fidelity, well, 12,500Hz is a good start. A loty of people never heard the flyback transformer on their old TVs vibrate, but I can hear them loud and clear. That's 15,750Hz.
And AM radio can sound very, very good. AM in America has a theoretical response of 16KHz, but currently is restricted in the U.S. to 10.2KHz (since 1989) to accomodate more stations and reduced interference from distant stations. The BBC at one time sent good audio, and a few shortwave stations did, and old AM radios had great speakers because they sent pretty good audio back then. Reducing response is also a way to extend range, along with compression, limiting, and a few other tricks that degrade ausio quality greatly. But AM is now the province of talk and news, so it doesn't seem to matter. FM, of course, also uses those tricks, and the result is nasty sound quality. To a generation broguth up on 128kbps MP3s, this is not a great loss. I code my music for my players at 320K or any of the lossless formats. 128k sizzle drive me crazy. And most FM music stations use MP3s anyways, they are largely programmed nationally and delivered over a satellite link. Tragedy.
To ask for improved sound quality in telephone is to ask for some compromises - fewer conversations over a given link, fewer conversations per cell tower, more Internet bandwidth. I'm pretty sure none of the incumbents will bother, as this ultimately results in increased direct costs, and probably zero increased revenue. Skype, etc., play with the codec and give apparently better results, the emphasis on 'apparently'. There are some clever audio tricks that will give a more pleasing experience with very little increase in bandwidth. Maybe Android can play with the audio, but I bet Apple could care less. The ILECS, bah!
So, the legacy of telephony is an old one, and has left us with something that works, but not as well as it could. Just a few more dollars, and you could have better!
deleting the extra space after periods so i can stay relevant, yeah.
"3.3Khz was the accepted frequency that telephone calls are going to run"
The bandwidth was not "accepted". It was set by the engineers that design the first analog telephone systems. It is a compromise between the need to have very small bandwidth per channel (so you can multiplex a lot of channels, and send them on the expensive long-distance cable) and the need to understand what the other person is saying and also, very important, to recognize who that person is (large bandwidth is better). They made some tests and this is how they found the sweet spot.
Speex, motherfucker, do you use it?
I totally agree. I've basically had it with cell phone voice quality. Sometimes I use Skype and I usually say something like "Hey this sounds just like phones did back in like 1985" and I get all nostalgic for Star Wars, ALF, and (strangely) the Reagan administration.
I went most of this decade using a cell phone, and after my GF got a landline at her new apartment, I decided to get one at mine as well. I found that roughly 30% of the conversation on cell phones is one party asking the other "can you repeat that" due to the miserable fidelity. We both use VOIP landlines, which I can't directly compare with plain copper, but it is a dramatic improvement over cell phones. (Now our conversations are 30% shorter.)
I haven't had this problem on a wired phone in the US, even with digital signaling in between. The Phone Company actually went through a great deal of trouble to keep that working.
Well, I don't know about worse, but I've experienced latency like that on long-distance calls over landlines many times over the past 10 years or so.
Are you adequate?
I bet boffins did. But they were wrong. Serves them right for being boffins, I suppose.
It's bandwidth, not frequency. In the USA, POTS (Plain Old Telephone Service) lines are 3 KHz, specifically 400 Hz to 3.4 KHz. 400 Hz is the low frequency, and that is way above the lowest tones in most voices; while 3.4 KHz, the highest frequency passed, is way below the highest tones in most voices. But the reason for the choice was this range provides very good intelligibility -- that is, ease of understanding -- for almost all voices, and at the time, wider bandwidth meant more expensive components multiplied by a huge, and growing, phone system.
Basically, many nuances of speech were foregone as a matter of financial triage.
I've fallen off your lawn, and I can't get up.
The VOIP world has spent a lot of time arguing about codecs, and is a MOS score less than 3.9 adequate for toll quality, and the IP PBX business was having to convince customers that 8kbps G.729 codecs were good enough for business, you didn't need full 64kbps G.711 uncompressed voice. Fortunately, cell phones became universal a few years back, so customers got used to low-bandwidth sound, compression, and cheap little microphones with road noise and passing trucks in the background, and somehow the MOS scores just stopped mattering so much. At least most of us don't have passing trucks to deal with when we're at our desks, though the rack of routers behind me is annoyingly loud. The real problem has become how to avoid multiple rounds of codecs on calls between people on separately managed VOIP systems, especially if one's a mobile phone using GSM codecs and the other is a PBX using G.729 codecs, which do different kinds of damage to the voice signal.
However, Mr. AC, unless you're in China, my guess is that your third-world country is using GSM, so at best you'd be using one of the AMR codecs, which still start off by sampling the voice at 8k samples/sec, and are therefore limited to 4KHz audio, just like telco phones running on T1 or E1 lines. They may be using the better flavors of GSM codec at 12.2kbps, as opposed to 6.7kbps or 5.9 or whatever, but that's how much damage they've done to the sound after it was already digitized.
Bill Stewart
New Fast-Compression-only CPR http://preview.tinyurl.com/dy575ks
You haven't look at phone prices any time in the last 15 years if you think that's what most unsubsidized phones cost. The very highest-end, most expensive phones happen to cost around there. Sheesh, for $500 you can get an unsubsidized Nokia N900!
If you're going to throw words around like "the iPad makes a very cheap phone" then you need to look harder for some cheap phones. Truly cheap unsubsidized phones can be had for less than $60 (and I'm not even shopping around very hard here, I bet someone can find an older one for $20) and that's if you totally blow off the used market. Granted, these things aren't nearly as capable as an iPad or any other laptop computer, but you're the one who said "cheap" and used being-a-phone as the example application.
Enjoy your iPad. But please, drop the "makes a very cheap phone" nonsense because it simply isn't true.
We're seeing more G.722 in VOIP phone sets these days. This gives you 7 khz bandwidth which is respectable for voice. It's also a royalty free codec that's simple to implement. It's supported (mostly) in Asterisk and is commonly used by the corporate conference systems and radio stations. There are better codecs, but the royalties preclude their inclusion into the things that most people buy. Cell phones, as far as audio go, as a disease! I used to be the Chief Engineer at a major talk radio station and... dealing with cell phones was just awful.I refuse to participate on a conference call or any critical phone call using a cell phone. How people can use those things as their primary home phones mystifies me.
The FreeSWITCH developers are on an audio conference call all day long. Most of us use G.722 at 16k or G.722.1 at 32k. When someone calls in on a cell phone (GSM) or land line (PCMU, aka G.711u) the difference is more than remarkable. When you are on a nice headset in 16k (or higher) all day long then you begin to appreciate how horrible the legacy stuff really is. The sad part is that G.711 takes 64kbps for an audio signal at 8kHz. We do lots of codecs that are higher quality and use much less bandwidth. For example, we can get a single channel of 48kHz CELT audio in 64kbps. (If you have a nice headset and your partner does as well then you will find it almost eerie how crisp and clear the sound is!)
:) Interestingly, you can't hear the crickets on an 8k connection! Just thought I'd share that tidbit.
People who are content with G.711 and GSM need to ask themselves why. Is "good enough" really good enough? I've heard this from the Asterisk camp on more than one occasion: "You'll never need more than 8kHz audio!" I strongly disagree with that assertion. After using FreeSWITCH for the past few years I really appreciate the value of HD VoIP. I could not imagine telecommuting and being stuck on a crappy 8kHz connection all day, listening to other people on a crappy 8kHz connection.
If you haven't checked out HD VoIP then you owe it to yourself to see what's out there.
-MC
Just an interesting side note: in our conference we often play sound bites. We have a sound bite of crickets chirping - the sound that gets played when someone asks a question and no one answers.
The hell with Ventrillo, we switched to Mumble 6 months ago and would never go back
http://mumble.sourceforge.net/
Jonah HEX
Horror & SciFi Erotic Nudes
Is a boffin some kind of sea-going bird?
Maybe this should have been posted over on gibberish.slashdot.org
Boffins did not decide this. It is a consequence of the evolution of human speech, both vocalization and hearing. The part of our speech that encodes content is the part that telephones have been engineered to convey. It's actually less than the 300-3400 Hz band, there's a mostly-useless band segment between the low frequencies and the higher ones that can be left out without much effect on speech recognition.
There have been a number of "hi-fi" schemes for telephony and bandwidth-limited radio. Some add bass, which is really cheap to do in terms of bandwidth because there is only a narrow band to be added and there's not much real information there at all so that you can compress the heck out of it and it still sounds like speech. AMBE+ does this on two-way radio, with a rather irritating synthetic bass. The other is to add more highs, and then you are going to mostly get more sibilants.
It is going to end up working better on the music-on-hold than it will on real voice.
Bruce Perens.
Codec2 is a digital voice codec for ham radio and potentially all low-bandwidth voice communication. Currently it fits in 2550 bits per second, and we expect it to get narrower. See the Alpha Release Code.
Bruce Perens.
Nothing wrong with the POTS. They are practically now digitised from one end of the telephone exchange to the receiving end exchange. For long distance if call goes through sattelite and if it ended in two hops up to the sky, then the lag is noticeable but quite OK if you make the adjustment. But the call will still be clear because of the digital transmission except if there are echos. But now I don't know what are the routes taken by a call from a POTS, could be that it be routed through the internet, cheaper for the telcos. I can only assume.
I prefer the low quality, long distance phone calls, and the correspondingly low price that low quality offers me.
Well, although the local service I have here seems "ok" most of the time, after visiting overseas where they're still using full-analog for local calls (backwoods, India, for example), the quality is surprisingly better... and home seems crappy upon return.
My experience only... maybe I'm just getting old and can't handle duplex myself anymore. :)
I used to have a sig, but I set it free and it never came back.
The company I have been working for has been testing a wide range of VoIP handsets and what surprised us the most is even though the phones themselves can use a wideband codec (Siren 16 comes to mind) the actual handpiece mic design is primitive to say the least (not talking el cheapo digital telephones here but models around $300-600). A majority of the models we tested had a simple pin hole mic on the handpiece with no noise canceling at all (done in software I suspect). We often found that by just changing the handset for one with a good noise canceling mic within a well thought out internal cavity (yes even the internal shape of the handset effects the audio quality) improved the quality of the audio massively. We suspect the reason for this is because designers now think they can do everything in software, The reality doesnt match up so you have these expensive digital telephones with very well designed codecs but the hardware so badly designed audio wise that you end up with audio quality that is no better (or worse) than a analogue handset.
Also: Phone systems work on standards to exchange signals. A big one is the TDM CODECs.
For about a half century the telcos have been digitizing at 8,000 samples per second using one of two 8-bit codings called "A-law" and "U-law" (where "U" is actually "mu"). (Think 8-bit signed floating point numbers with smoothing tweaks around the values where the exponent changes.) 8k samples/sec has a Nyquist frequency limit of 4,000 Hz (and you have to low-pass filter it somewhat below that to keep higher frequencies from being "folded back" around 4kHz and fouling the signal you're after.
While digital cellphones and VoIP have been driving adoption of other CODECs, the push has been to reduce the data rate required to carry a "voice-quality" call (either on wires or through radio noise on limited cellular bandwidth) and to survive transcoding to and from A/U-law for interchange with the telcos' installed base, rather than taking advantage of higher wire/fiber data rates to improve audio fidelity.
Bantam Dominique roosters crow a four-note song. Once you've heard it as "Happy BIRTHday" you can't NOT hear it that way
In the 19th century somebody decided that QWERTY is the best layout for a keyboard. Nowadays even some cellphones hava a QWERTY layout...
...when the earpieces on phones these days sound like a Marshall stack on 10 (very loud guitar amp) from about 3 miles away? The frequency response is probably somewhere around a spike between 500 and 2k Hz, with next to nothing at all above or below that...
OK, Sipdroid or Skype sounds a bit better than regular phone calls, but the problem really is the earpiece.
Here's a suggestion for manufacturers: Make the screen 1cm shorter, and use the increased space to put in MULTIPLE (I'm thinking three next to each other) speakers with AT LEAST a 10mm membrane in there. That way it'll sound great at low volume during calls, and there's no need to have the speakers for speakerphone/video etc. on the back (which is the stupidest idea ever, by the way... blaring right into the palms of my hands is oh so very efficient).
My HTC Desire is especially bad, but even phones with supposedly decent earpieces sound like shit.
I notice this very much when I am talking to certain people, but I think it is an issue with the telephone handset, not with the line (for landlines at least).
I assumed the cause is the feedback canceling mechanism in the handset. As handsets have become smaller, manufacturers have had to resort to more extreme measures to stop the microphone picking up the output of the speaker.
Some handsets seem to completely turn off the speaker whenever the microphone is picking up sound. This effectively means that the person talking into that handset has no way to know if the caller at the other end is trying to interject a comment or even interrupt.
This shouldn't be a problem, except that certain people seem to be almost irrationally uncomfortable with any moment of silence on a phone line, and so will just talk incessantly unless they know the person at the other end is trying to interrupt.
If you couple one of these people with such a feedback canceling handset, you end up with an enforced monologue (possibly followed by complaints that the other person never tells them anything).
Back in the 1980s we were doing some work on tank internal networking. Our group turned up on Salisbury Plain for a trial on a hot, dry day. Whereupon I got my all-in-one biker suit and balaclava out of my kit bag just as my colleagues realised they were about to get their suits covered in dust. At which the sergeant in charge of the support crew remarked "You know, Sir, you're quite sensible for a boffin". I asked him if he'd let me quote him on my CV.
From scarped cliff or quarried stone she cries "A thousand types are gone, I care for nothing, no not one."
In WW2, instead of protecting him like Churchill, he was allowed to go up in bombers - and was killed. Our deeply stupid Civil Service, then as now, has no notion of the value of scientists.
From scarped cliff or quarried stone she cries "A thousand types are gone, I care for nothing, no not one."