Е НЕЕЕЕ....ТЕЯ ВЕЧЕ ПЕЧЕЛЯТ НАГРАДАТА СМЕШНИЦИ НА ХИЛЯДОЛЕТИЕТО!!!!

Това нещо...го пише във всяка една книга...или в материали за GSM стандарта в нета !!!!

Ето:

SPEECH CODING

GSM is a digital system, so speech, which is inherently analog, has to be digitised. 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. 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 (ms) samples, each of which is encoded as 260 bits, giving a total bit rate of 13Kbps (kilobits per second). 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 13Kbps bit rate.

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The overall data rate for the radio channel is 270kb/s.
This is split into B full rate or 16 half rate traffic channels, plus the signaling channels.
The coding is complex in order to have the maximum chance to detect and correct the errors encountered in a typical propagation path.
The output of the speech coder is encrypted, coded and interleaved in a sophisticated way to allow Forward Error Correction to be applied.
The data is then sent as bursts in time slots of 577µs, each containing 116 encrypted bits.
There are 8 or 16 of these time slots per TDMA frame, and the receive and transmit time slots are staggered so that the mobile station is not receiving at the same instant as it is transmitting, thus simplifying the filtering requirements.
With this scheme, there can be at least one spare slot between transmit and receive, leaving time for the synthesizer to change frequency (whether or not hopping is employed).
The receiver also monitors adjacent cell for one time slot each frame to determine their signal strength to optimize a possible handover,

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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.

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.

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Channel coding and modulation
Because of natural and man-made electromagnetic interference, the encoded speech or data signal transmitted over the radio interface must be protected from errors. GSM uses convolutional encoding and block interleaving to achieve this protection. The exact algorithms used differ for speech and for different data rates. The method used for speech blocks will be described below.

Radio waves are a rough medium to transmit fragile data over; we need a way to protect that information. We do so with error checking, mathematical routines that check and then double-check the integrity of our data. These methods contribute greatly to the overhead in a digital stream, adding a tremendous amount of bits, and thus dramatically cutting down on data speed. It's one reason data transfer rates are only 9.6kbs. This is a complex subject, one I haven't written much on.

Recall that the speech codec produces a 260 bit block for every 20 ms speech sample. From subjective testing, it was found that some bits of this block were more important for perceived speech quality than others. The bits are thus divided into three classes:

Class Ia 50 bits - most sensitive to bit errors
Class Ib 132 bits - moderately sensitive to bit errors
Class II 78 bits - least sensitive to bit errors
Class Ia bits have a 3 bit Cyclic Redundancy Code added for error detection. If an error is detected, the frame is judged too damaged to be comprehensible and it is discarded. It is replaced by a slightly attenuated version of the previous correctly received frame. These 53 bits, together with the 132 Class Ib bits and a 4 bit tail sequence (a total of 189 bits), are input into a 1/2 rate convolutional encoder of constraint length 4. Each input bit is encoded as two output bits, based on a combination of the previous 4 input bits. The convolutional encoder thus outputs 378 bits, to which are added the 78 remaining Class II bits, which are unprotected. Thus every 20 ms speech sample is encoded as 456 bits, giving a bit rate of 22.8 kbps.

To further protect against the burst errors common to the radio interface, each sample is interleaved. The 456 bits output by the convolutional encoder are divided into 8 blocks of 57 bits, and these blocks are transmitted in eight consecutive time-slot bursts. Since each time-slot burst can carry two 57 bit blocks, each burst carries traffic from two different speech samples.

Recall that each time-slot burst is transmitted at a gross bit rate of 270.833 kbps. This digital signal is modulated onto the analog carrier frequency using Gaussian-filtered Minimum Shift Keying (GMSK). GMSK was selected over other modulation schemes as a compromise between spectral efficiency, complexity of the transmitter, and limited spurious emissions. The complexity of the transmitter is related to power consumption, which should be minimized for the mobile station. The spurious radio emissions, outside of the allotted bandwidth, must be strictly controlled so as to limit adjacent channel interference, and allow for the co-existence of GSM and the older analog systems (at least for the time being).


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