Content for TS 46.002 Word version: 16.0.0
The present document is an introduction to GSM 06.20 
, GSM 06.21 
, GSM 06.22 
, GSM 06.41 
and GSM 06.42 
TSs dealing with the speech processing functions in the half rate channel of the GSM system. A general overview of the speech processing functions is given, with reference to the TSs where each function is specified in detail.
The following documents contain provisions which, through reference in this text, constitute provisions of the present document.
References are either specific (identified by date of publication, edition number, version number, etc.) or non specific.
For a specific reference, subsequent revisions do not apply.
For a non-specific reference, the latest version applies. In the case of a reference to a 3GPP document (including a GSM document), a non-specific reference implicitly refers to the latest version of that document in the same Release as the present document.
GSM 01.04: "Digital cellular telecommunication system (Phase 2+); Abbreviations and acronyms".
GSM 03.50: "Digital cellular telecommunication system (Phase 2+); Transmission planning aspects of the speech services in the GSM Public Land Mobile Network (PLMN) system".
GSM 05.03: "Digital cellular telecommunications system (Phase 2+); Channel coding".
GSM 06.20: "Digital cellular telecommunications system (Phase 2+); Half rate speech: Half rate speech transcoding".
GSM 06.21: " Digital cellular telecommunications system (Phase 2+); Half rate speech; Substitution and muting of lost frame for half rate speech traffic channels".
GSM 06.22: "Digital cellular telecommunications system (Phase 2+); Half rate speech; Comfort noise aspects for half rate speech traffic channels ".
GSM 06.41: "Digital cellular telecommunications system (Phase 2+); Half rate speech; Discontinuous transmission (DTX) for half rate speech traffic channels".
GSM 06.42: "Digital cellular telecommunications system (Phase 2+); Half rate speech; Voice Activity Detector (VAD) for half rate speech traffic channels ".
ITU T Recommendation G.711: "Pulse Code Modulation (PCM) of voice frequencies".
GSM 06.07: "Digital cellular telecommunications system (Phase 2+); Half rate speech; Test sequences for the GSM half rate speech codec".
GSM 06.06: "Digital cellular telecommunications system (Phase 2+); Half rate speech; ANSI C code for the GSM half rate speech codec".
Definition of terms used in the present document can be found in GSM 06.20 
, GSM 06.21 
, GSM 06.22 
, GSM 06.41 
and GSM 06.42 
For the purposes of the present document, the following abbreviations apply.
Bad Frame Indication
Base Station System
Global System for Mobile communications
Pulse Code Modulated
Public Land Mobile Network
Public Switched Telephone Network
Slow Associated Control CHannel
Time Alignment Flag
Vector Sum Excited Linear Predictor
Unreliable Frame Indication
For abbreviations not given in this subclause, see GSM 01.04 
presents a reference configuration where the various speech processing functions are identified. In this figure, the relevant documents for each function are also indicated.
In Figure 1
, the audio parts including analogue to digital and digital to analogue conversion are included, to show the complete speech path between the audio input/output in the Mobile Station (MS) and the digital interface of the Public Switched Telephone Network (PSTN). The detailed specification of the audio parts are contained in GSM 03.50 
. These aspects are only considered to the extent that the performance of the audio parts affect the performance of the speech transcoder.
Figure 1: Overview of audio processing functions
The half rate speech transcoding function is described in GSM 06.20 
As shown in figure 1, the speech encoder takes its input as a 13 bit uniform Pulse Code Modulated (PCM) signal either from the audio part of the MS or on the network side, from the PSTN via an 8 bit/A law or μ-law (PCS 1900) to 13 bit uniform PCM conversion. The encoded speech at the output of the speech encoder is delivered to the channel coding function as defined in GSM 05.03 
to produce an encoded block consisting of 228 bits leading to a gross bit rate of 11,4 kbit/s.
In the RX direction, the inverse operations take place.
GSM 06.20 
describes the detailed mapping between input blocks of 160 speech samples in 13 bit uniform PCM format into encoded blocks of 112 bits and from encoded blocks of 112 bits to output blocks of 160 reconstructed speech samples. The sampling rate is 8 000 sample/s leading to an average bit rate for the encoded bit stream of 5,6 kbit/s. The coding scheme is called Vector Sum Excited Linear Prediction (VSELP) coding.
GSM 06.20 
describes the codec and GSM 06.06 
defines the C code, thus enabling the verification of compliance to GSM 06.20  to a high degree of confidence by use of a set of digital test sequences given in GSM 06.07 
The half rate discontinuous transmission function is described in GSM 06.41 
During a normal conversation, the participants alternate so that, on the average, each direction of transmission is occupied about 50 % of the time. Discontinuous Transmission (DTX) is a mode of operation where the transmitters are switched on only for those frames which contain useful information. This may be done for the following two purposes:
in the MS, battery life will be prolonged or a smaller battery could be used for a given operational duration;
the average interference level over the air interface is reduced, leading to better Radio Frequency (RF) spectrum efficiency.
The overall DTX mechanism is implemented in the DTX handlers (Transmit (TX) and Receive (RX)) described in GSM 06.41 
and requires the following functions:
a Voice Activity Detector (VAD) on the TX side, see GSM 06.42 ;
evaluation of the background acoustic noise on the TX side, in order to transmit characteristic parameters to the RX side, see GSM 06.22 ;
generation of comfort noise on the RX side during periods where the radio transmission is turned off, see GSM 06.22 .
The transmission of comfort noise information to the RX side is achieved by means of a SIlence Descriptor (SID) frame. The SID frame is transmitted at the end of speech bursts and serves as an end of speech marker for the RX side. In order to update the comfort noise characteristics at the RX side, SID frames are transmitted at regular intervals also during speech pauses. This also serves the purpose of improving the measurement of the radio link quality by the Radio SubSystem (RSS).
The DTX handlers interwork with the RSS using flags. The RSS is controlled by the transmitter keying on the TX side, which performs pre processing functions on the RX side. This is described in GSM 06.41 
The speech flag (SP) indicates whether information bits are speech or SID information. The SP flag is calculated from the VAD flag by the TX DTX handler. When SID information is transmitted (SP="0") the operation of the speech encoder is modified to reduce the remaining computation for that frame. This is described in GSM 06.22 
The half rate VAD function is described in GSM 06.42 
The input to the VAD is a set of parameters computed by the half rate speech encoder defined in GSM 06.20 
. The VAD uses this information to decide whether each 20 ms speech coder frame contains speech or not.
GSM 06.42 
describes the VAD algorithm and GSM 06.06 
defines the C code. The verification of compliance to GSM 06.42 
is achieved by use of digital test sequences (see GSM 06.07 
) applied to the same interface as the test sequences for the speech codec.
The half rate noise comfort insertion function is described in GSM 06.22 
When switching the transmission on and off during DTX operation, the effect would be a modulation of the background noise at the receiving end, if no precautions were taken. When transmission is on, the background noise is transmitted together with the speech to the receiving end. As the speech burst ends, the connection is off and the perceived noise would drop to a very low level. This step modulation of noise may be perceived as annoying and reduce the intelligibility of speech, if presented to a listener without modification.
This "noise contrast effect" is reduced in the GSM system by inserting an artificial noise, termed comfort noise, at the receiving end when speech is absent.
The comfort noise processes are as follows:
the evaluation of the acoustic background noise in the transmitter;
the noise parameter encoding (SID frames) and decoding;
and the generation of comfort noise in the receiver.
The comfort noise processes and the algorithm for updating the noise parameters during speech pauses are defined in detail in GSM 06.22 
The comfort noise mechanism is based on the half rate speech codec defined in GSM 06.20 
The half rate speech frame substitution and muting function is described in GSM 06.21 
In the receiver, frames may be lost due to transmission errors or frame stealing.
GSM 06.21 
describes the actions to be taken in these cases, both for lost speech frames and for lost SID frames in DTX operation.
In order to mask the effect of an isolated lost frame, the lost speech frame is substituted by a predicted frame based on previous frames. Insertion of silence frames is not allowed. For several subsequent lost frames, a muting technique shall be used to indicate to the listener that transmission has been interrupted.
The GSM half rate speech transcoder, VAD, DTX system and comfort noise parts of the audio processing functions (see figure 1) are defined in bit exact arithmetic. Consequently, they shall react on a given input sequence always with the corresponding bit exact output sequence, provided that the internal state variables are also always exact in the same state at the beginning of the experiment.
The input test sequences provided in GSM 06.07 
shall force the corresponding output test sequences, provided that the tested modules are in their home state when starting.
The modules may be set into their home states by provoking the appropriate homing functions.
Special inband signalling frames (encoder homing frame and decoder homing frame described in GSM 06.20 
have been defined to provoke these homing functions also in remotely placed modules.
This mechanism is specified to support three main areas:
type approval of mobile terminal equipment;
type approval of infrastructure equipment;
remote control and testing for operation and maintenance.
At the end of the first received homing frame, the audio functions that are defined in a bit exact way shall go into their predefined home states. The output corresponding to the first homing frame is dependent on the codec state when the frame was received. Any consecutive homing frames shall produce corresponding homing frames at the output.