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Thomas Farely

Tom has produced privateline.com since 1995. He is now a freelance technology writer who contributes regularly to the site.

His knowledge of telecommunications has served, most notably, the American Heritage Invention and Technology Magazine and The History Channel.
His interview on Alexander Graham Bell will air on the History Channel the end of 2006.

Ken Schmidt

Ken is a licensed attorney who has worked in the tower industry for seven years. He has managed the development of broadcast towers nationwide and developed and built cell towers.

He has been quoted in newspapers and magazines on issues regarding cell towers and has spoke at industry and non-industry conferences on cell tower related issues.

He is recognized as an expert on cell tower leases and due diligence processes for tower acquisitions.

« Burst structure | | Channel coding and modulation »

January 17, 2006

Posted by Tom Farley & Mark van der Hoek at 11:58 AM

Speech coding

Speech coding means turning voice into digital. I've written much on this subject so be sure to click on the links below if there are points you don't understand . . .

GSM is a digital system, so speech which is inherently analog, has to be digitized. The method employed by ISDN, and by current telephone systems for multiplexing voice lines over high speed trunks and optical fiber lines, is Pulse Coded Modulation (PCM). The output stream from PCM is 64 kbps, too high a rate to be feasible over a radio link. The 64 kbps signal, although simple to implement, contains much redundancy. The GSM group studied several speech coding algorithms on the basis of subjective speech quality and complexity (which is related to cost, processing delay, and power consumption once implemented) before arriving at the choice of a Regular Pulse Excited -- Linear Predictive Coder (RPE--LPC) with a Long Term Predictor loop.

Conventional cellular uses an equally intimidating algorithm named Vector Sum Excited Linear Predictive speech compression. Ugh. Click here to learn about it.

Basically, information from previous samples, which does not change very quickly, is used to predict the current sample. The coefficients of the linear combination of the previous samples, plus an encoded form of the residual, the difference between the predicted and actual sample, represent the signal. Speech is divided into 20 millisecond samples, each of which is encoded as 260 bits, giving a total bit rate of 13 kbps.

This is the subject of digital signal processing. Read about it here.

This is the so-called Full-Rate speech coding. Recently, an Enhanced Full-Rate (EFR) speech coding algorithm has been implemented by some North American GSM1900 operators. This is said to provide improved speech quality using the existing 13 kbps bit rate.

Nokia said in January, 1997 that they would start shipping Enhanced Full Rate voice codecs by March 1997: http://press.nokia.com/PR/199701/775480_5.html; I must assume their use is now wide spread.

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