|
|
Audio transmission in a GSM network
Deze tekst is alleen in het Engels beschikbaar. Omdat vrijwel alle termen ook engels zijn zou een vertaling een warboel opleveren.
Speech
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. 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 so-called Full-Rate (FR) speech coding. An Enhanced Full-Rate (EFR) speech coding algorithm has been implemented to provide improved speech quality using the existing 13 kbps bit rate.
Channel coding
Because of 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. Tthe 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. This is comparable with the LSB from digital images what JPG compression is based on. The
bits of the audiosample are 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
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.
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.
|
|
|
