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Communications

Lo-Fi Phones and the Future 228

Posted by CmdrTaco
from the who-needs-it dept.
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.
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Lo-Fi Phones and the Future

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  • 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.
    • The Coral Cache version works.

      • Is it just me, or does anyone else think it'd be somewhat wise to have Slashdot story submission code automatically turn links into Coral Cache links?

        • Re: (Score:3, Funny)

          by fyngyrz (762201)

          It's just you. We don't read the TFA; the only reason we click on those links is we like to crush people's servers. Get with the program, please.

        • Is it just me, or does anyone else think it'd be somewhat wise to have Slashdot story submission code automatically turn links into Coral Cache links?

          There's a Firefox plugin [mozilla.org] that does this for you.

          Ignore the "Not available for Firefox 3.6" message - you can override compatibility checking with the Add-on Compatibility Reporter [mozilla.org] plugin and it works just fine.

  • guess what (Score:4, Informative)

    by Anonymous Coward on Wednesday September 08, 2010 @02:32PM (#33512292)

    I live in 3rd world country and our major cellphone networks support hd-voice codecs.

  • Bandwidth (Score:3, Funny)

    by DIplomatic (1759914) on Wednesday September 08, 2010 @02:32PM (#33512298) Journal
    Looks like their server has Lo-Fi bandwidth....
  • Otherwise known as... Slashdotted. I hope their ISP doesn't put the hammer down on them.
  • 503. (Score:2, Informative)

    by mm_202 (1569029)
    Slashdotted already?
    hmm, well it is running IIS...
  • Coral Cache .... (Score:5, Informative)

    by Qubit (100461) on Wednesday September 08, 2010 @02:34PM (#33512340) Homepage Journal

    Just click here [nyud.net] and avoid the Slashdotting...

    • by Unbeliever (35305)

      Until the Coral Cache caches the "We've been slashdotted" page.

    • 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 ba

  • Sorry Taco, I disagree - I've yet to be told during a conference calls to go handle "many whelps."
  • by RichMan (8097) on Wednesday September 08, 2010 @02:35PM (#33512352)

    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.

    • by grumpyman (849537)
      Ha... if 3.3KHz was the accepted frequency all they'll hear probably sounds like bird chirping.
      • by TeknoHog (164938)

        Ha... if 3.3KHz was the accepted frequency all they'll hear probably sounds like bird chirping.

        No, that 3.3 kHz would be a constant sine wave. Turning that sound on and off requires some bandwidth.

    • by avandesande (143899) on Wednesday September 08, 2010 @02:57PM (#33512624) Journal

      Damn it, 3.3Khz is more Khz than anyone will ever need!

    • by Gordonjcp (186804) on Wednesday September 08, 2010 @03:10PM (#33512816) Homepage

      Actually the bandwidth is 3.1kHz, running from 300Hz to 3.4kHz. This is the range of frequencies that conveys the most information relevant to intelligibility. Anything else makes it easier to recognise the speaker but doesn't make it easier to understand them and can make it harder to understand in noisy environments.

      If you low-pass filter speech below 3.4kHz then mostly you only lose the high frequencies in sibilants, but you also filter out a lot of background noise. If you're really interested you could set up a media player to play recorded speech through a tunable bandpass filter and see what you can filter out before the speech becomes hard to understand. Once you've got a feel for how the filters affect the intelligibility, try playing it back in a noisy environment (or mix in a recording of the inside of a car or something).

      The 300-3400Hz filter is pretty standard in communications, and it crops up in all sorts of places. I wrote a software-defined radio app that defaults to 300-3400 but is easily tunable up to 15kHz for either low or highpass (although if you highpass filter at 15kHz you won't hear much). Occasionally I'll use it to roll off above about 2.8kHz to remove high-pitched squeaky noises from adjacent transmitters.

      Obligatory screenshot: http://www.gjcp.net/~gordonjcp/lysdr.jpg [gjcp.net]

      You can see the yellow strip representing the passband of the filter. The (fairly weak) signal shown doesn't really have any strong components much above about 2kHz, and I could reduce the noise by sliding the leftmost (lowpass) filter in a bit. A quick explanation of what you're looking at - that's a spectrum plot of a chunk of the 7MHz amateur band, with lower frequencies on the left as indicated by the scale at the bottom. Since on 7MHz we use lower sideband (LSB) [wikipedia.org], the higher audio frequencies correspond to lower RF frequencies (further away from the red tuning cursor).

      • by jmv (93421) on Wednesday September 08, 2010 @07:04PM (#33515402) Homepage

        Actually the bandwidth is 3.1kHz, running from 300Hz to 3.4kHz. This is the range of frequencies that conveys the most information relevant to intelligibility. Anything else makes it easier to recognise the speaker but doesn't make it easier to understand them and can make it harder to understand in noisy environments.

        That's a myth, 3.4 kHz is not high enough to tell the "f" sound from the "s" sound over the phone. Similarly for "v" vs "z" and a bunch of others. If phones were that intelligible, people wouldn't have to say "a as in alpha, b as in bravo, ...".

    • Another thing is that while Skype calls can be better, they're often among the worst that I've received too. If you're doing it with the $1 mic that came with the computer, then you're probably doing your recipient a disfavor. It isn't about Skype per se, it's a matter of where the weakest link is, even a crappy phone seems to have a more tolerable mic pickup than standard computer mics.

    • by TeknoHog (164938)
      My thoughts exactly. One problem is that non-technical people tend to use "bandwidth" as a synonym for channel capacity (data transfer rate), instead of its original meaning (the width of a range of frequencies). So perhaps it is safer to avoid using that word altogether.
  • 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.

  • Latency? (Score:4, Interesting)

    by Moridineas (213502) on Wednesday September 08, 2010 @02:40PM (#33512392) Journal

    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??

    • My vonage service is indistinguishable from a copper connection.

    • 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?

      Depends on your connection speed. Sometimes when playing a game that taxes my computer & Internet connection I can say something in Vent and then ~1 min later when someone else says something, I can hear my comments playing in the background.

      So it is possible to have latency issues.
    • Re: (Score:3, Interesting)

      by Steve Max (1235710)

      My only landline is a VOIP service from my cable provider (Net, from Brazil). There are some downsides (like the fact that the line goes down when the power does, or the time it takes from the moment you switch on the adapter to actually getting a dial tone, which is a problem when you return home from a trip), but I've NEVER experienced a voice delay. YMMV, of course, but Net allocate a fixed bandwidth to the voice service above the amount you have for your Internet connection; and they give voice a much h

    • Re: (Score:3, Interesting)

      by martas (1439879)
      packet switching is always going to have higher latency than circuit switching... pretty much inevitable. so no, it's not just you.
      • by epine (68316)

        packet switching is always going to have higher latency than circuit switching... pretty much inevitable. so no, it's not just you.

        I have a telecoms book which explains that at a certain point in the past, calls early in the morning EST in the oversubscribed east coast corridor (which I recall as Boston to Washington) were sometimes circuit switched via California, since the east-west links had a lot of excess capacity at 05:00 PST. This strategy sucked for the phone company if the call lasted more than th

    • I've gone from land line to Vonage to running Asterisk with several VOIP gateway services, which I've been doing for about 5 years. VOIP quality primarily depends on routing. I've currently got a VOIP gateway provider with ping times 40 ms and the latency and jitter during calls is not a problem and definitely better than a cell phone. I can't control how the gateway routes calls and it seems that some area codes I call have some noticeable latency, but overall it is a good experience. I've had many pe

    • Re:Latency? (Score:4, Informative)

      by natehoy (1608657) on Wednesday September 08, 2010 @03:41PM (#33513256) Journal

      Latency of the call is highly variable, and dependent on two factors:

      1. How much latency is in the network?
      2. How much latency is introduced by the VoIP conversion itself?

      I joined Vonage about 5 years ago. On my first ISP, I got a little over 3/4 second of latency on a really good ISP connection. This was annoying, but not enough to really make me want to spend two and a half times as much for a landline with a non-portable number. Eventually, Vonage went through a stretch of upgrades to the firmware on my adapter and the latency dropped to about 1/4 second (all but unnoticeable). However, I traveled a lot a couple of years later and found that hotel connections tend to have a lot more latency, so I got a cheap prepaid cell for when I was on the road. I settled down to a local job again and had a lot of trouble with my new ISP for a while, resulting in poor call quality and very high latency, then we got that straightened out and I was back to 1/4-second delay, which was pretty much the rule until my company issued me a cell phone with unlimited minutes, so I ditched my Vonage line because I didn't use it. But friends who have joined since have reported very low delays, almost unnoticeable, as long as their connections were good.

      So the technology has improved, but you are still dependent on someone who gives you the better tech, and on a good Internet connection between you and the adapter on the other end where the call is bridged back to a POTS network.

      However, landlines have a few features that people have a hard time giving up. Whether you are willing to pay for them is a different matter.

      1. No need to manage power to a device. If the wires are up, the connection is the telco's responsibility.

      2. Real, honest 911 with pretty much 100% accurate location awareness. Your tax dollars at work (which are a generous chunk of the difference between telco and VoIP).

      3. "Feedback loop" (you can hear yourself talk in your earpiece). This helps regulate your volume, which is why people tend to talk louder into cellphones (they don't get that feedback).

      4. No-delay talking. When telcos use VoIP, they use really high-end gear and fast networks to support it.

      5. True DTMF support. This has gotten a little better, but VoIP for the most part can't carry DTMF tones to sufficient clarity, so your local VoIP adapter has to recognize an attempt at one and generate a fresh tone that your analog gear can recognize. Conversations with people can occasionally be interrupted by a "BEEP" as your VoIP adapter misidentifies a sound in their voice or the background as a DTMF tone and faithfully reproduces it, and you may get occasional complaints of the same issue on the other side . If it fails to reproduce when needed and you run a menu system, your customers will really hate traversing your menus.

      The net result of all of that is, well, you get what you pay for. Telcos are expensive, but you are pretty much guaranteed a good call every time. Most of the gear you probably own was built to analog specifications, and the telcos are good at maintaining that spec.

      For most of us, cell or VoIP is sufficient. We're OK with slight delays, a less-than-perfect reproduction of our voices, the occasional errant DTMF tone, etc.

      If you run a business and you strongly feel that clear telephony is a vital part of your business, then it's probably worth paying for in your case, or at least paying for a REALLY good Internet connection and high-end VoIP gear, not consumer-grade stuff. Though you could always run one VoIP line for a while and see how it works out (just use it for less critical calls to start with).

      • by Nethead (1563)

        3. "Feedback loop" (you can hear yourself talk in your earpiece). This helps regulate your volume, which is why people tend to talk louder into cellphones (they don't get that feedback).

        The term you are looking for is sidetone.

      • The net result of all of that is, well, you get what you pay for. Telcos are expensive, but you are pretty much guaranteed a good call every time. Most of the gear you probably own was built to analog specifications, and the telcos are good at maintaining that spec.

        For most of us, cell or VoIP is sufficient. We're OK with slight delays, a less-than-perfect reproduction of our voices, the occasional errant DTMF tone, etc.

        Describing Wal-Mart's business model to a 'T' - you're willing to put up with almost any

    • Regular telephone systems are packet-switched behind the scenes these days, so latency is a function of the quality of your connection, true for both your phone company and your VOIP service.
    • Re:Latency? (Score:5, Informative)

      by Mr 44 (180750) on Wednesday September 08, 2010 @03:58PM (#33513474)

      An MIT student did his thesis on Voice vs. Data lantency on cell phones, you might be interested in his methodology and results:
      Quality of Service Analysis for Audio over Cellular Voice Networks [mit.edu]

    • by ceoyoyo (59147)

      Have you tried it any time recently? I haven't had a noticeable delay using Skype on a notebook in ages. There used to be a delay when I was using Fring with an iPhone 3G, but the official Skype app also works with no noticeable delay.

    • by AaronW (33736)

      My experience with Comcast VOIP has been I cannot tell the difference except that, if anything, the sound is clearer than POTS. I don't notice any significant latency. It is difficult to beat POTS for latency, though. For voice they typically send a burst of compressed voice data every 10 or 20ms. POTS typically uses a dedicated very lot latency network which will have a lower latency than packet switched networks.

  • by 0racle (667029) on Wednesday September 08, 2010 @02:41PM (#33512410)
    So how many boffins died to bring 3.3Khz to our phones?
    • So how many boffins died to bring 3.3Khz to our phones?

      74 years ago when they made that policy? I'd say probably all of them.

  • by Surt (22457) on Wednesday September 08, 2010 @02:43PM (#33512440) Homepage Journal

    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 saves your data? (Score:2, Interesting)

    by rmdyer (267137)

    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..."

    • Some of the calls on Ventrilo Harassment were taking place on "corporate vents", which despite that were for some reason wide open.

  • 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.

  • by NixieBunny (859050) on Wednesday September 08, 2010 @02:46PM (#33512482) Homepage
    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 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.

  • by guruevi (827432) <eviNO@SPAMsmokingcube.be> on Wednesday September 08, 2010 @02:50PM (#33512526) Homepage

    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.

  • by Drakkenmensch (1255800) on Wednesday September 08, 2010 @02:52PM (#33512548)

    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.

  • Right on the spot (Score:5, Informative)

    by Gruturo (141223) on Wednesday September 08, 2010 @02:56PM (#33512614)

    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 [sourceforge.net], 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)

    • by BobMcD (601576)

      By the way, if you like Ventrilo, try Mumble [sourceforge.net], 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)

      I absolutely LOVE Mumble, but two points...

      1) Vent has normalization also, you just have to Google how to do it.

      2) None of your pugs will use Mumble. Period.

      It genuinely is a superior product, but would be greatly improved if the server would be so kind as to interface with a Ventrilo client. Ease of transition, and all that.

      • by Gruturo (141223)

        I absolutely LOVE Mumble, but two points...

        1) Vent has normalization also, you just have to Google how to do it.

        Yep, I did, and it didn't work well enough for me even after fiddling with the values. I ended up having to adjust almost every single speaker, and everytime they reinstalled their pc / changed their nick I had to do it again.

        2) None of your pugs will use Mumble. Period.

        eek, pugs :)

  • It's a real problem. (Score:4, Informative)

    by Animats (122034) on Wednesday September 08, 2010 @03:01PM (#33512670) Homepage

    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.

    • Re: (Score:2, Informative)

      by Anonymous Coward

      ISDN voice never caught on in the US, but it's widely used in some European countries.

      It's also widely used for telephone interviews on TV/radio, due to the improved clarity.

      • Re: (Score:2, Interesting)

        by N1EY (817702)
        ISDN is also used for link lines. I know of one clear channel station that does not have good LOS between the studio and the transmitters. The have a ISDN line to transfer the audio.
  • 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 [wikipedia.org]".

  • by s52d (1049172) on Wednesday September 08, 2010 @03:22PM (#33512998)

    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

  • by MadCow42 (243108) on Wednesday September 08, 2010 @03:34PM (#33513174) Homepage

    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.

    • by six11 (579)

      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

    • by russotto (537200)

      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.

      I haven't had this problem on a wired phone in the US, even with digital signaling in between. The

  • "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?"

    • Re: (Score:3, Interesting)

      by RevWaldo (1186281)
      Hear, hear! Back in the day in the USA, before cellular, VoIP, and cordless phones, and when every home or office had at least one Model 2500 phone, no one ever complained about sound quality. Real mikes, real speakers, real bells, real buttons with springs in them, and a corded handset that could double as a weapon in a pinch. Sigh. Good times, good times.

      .
      • by jburroug (45317)

        And don't forget that you could use the handset to beat an attacker to death without dropping the call you were on!

  • by rickb928 (945187) on Wednesday September 08, 2010 @03:40PM (#33513242) Homepage Journal

    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!

  • engineered (Score:2, Informative)

    by vacarul (1624873)
    "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 s
  • by EEPROMS (889169) on Wednesday September 08, 2010 @08:08PM (#33515800)
    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.

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