9.gsm Speech & Channel Coding
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GSM Presentation7 Speech and Channel Coding
Channel Coding
The following figure shows the steps involved to transform speech audio to radio waves and vice versa.
GSM Speech Processing Steps
Speech compressed using a predictive coding scheme Divided into blocks, each of which is protected partly by cyclic code and partly by a convolutional code Interleaving to protect against burst errors Encryption for providing privacy Assembled into time slots Modulated for analog transmission using GMSK
Speech coding
The GSM speech codec that transforms the analog signal (voice) into a digital representation, must meet the following criteria: Maintain speech quality. Reduce redundancy in voice utterances. This reduction is essential due to transmission capacity limitation on the data channel. Adopt low complexity speech codec to reduce production costs.
GSM TRANSMISSION PROCESS
STAGE 1: ANALOG TO DIGITAL (A/D) CONVERSION STAGE 2: SEGMENTATION STAGE 3: SPEECH CODING STAGE 4: CHANNEL CODING STAGE 5: INTERLEAVING STAGE 6: CIPHERING/ENCRYPTION STAGE 7: BURST FORMATTING STAGE 8: MODULATION & TRANSMISSION
Speech Coding
In order to send the voice information across a radio network, first thing to be done is to turn the voice into a digital signal. GSM uses a method called RPE-LTP (Regular Pulse Excited - Long Term Prediction) with Linear Predictive Coding to turn our analog voice into a compressed digital equivalent. One of the primary functions of an MS is to convert the analog speech information into digital form for transmission using a digital signal. The analog to digital (A/D) conversion process outputs a collection of bits: binary ones and zeros which represent the speech input.
In modern phone systems, digital coding is used. The electrical variations induced into the microphone are sampled and each sample is then converted into a digital code. The voice waveform sampled at a rate of 8 kHz and sample is converted into an 8 bit binary number, representing 256 distinct values . Since we sample 8000 times per second and each sample is 8 binary bits, we have a bitrate of 8kHz X 8 bits = 64kbps.
This bitrate is unrealistic to transmit across a radio network. GSM speech coding works to compress the speech waveform into a sample that results in a lower bitrate using RPE-LTP. The speech signal is divided into blocks of 20ms . Once we have a digital signal we have to add some sort of redundancy so that we can recover from errors when we transmit our digital voice over the radio channel.
These blocks are then passed to the speech codec of 13 kbps, to obtain speech frames of 260 bits each.
GSM Channel Coding
Once we have a digital signal we have to add some sort of redundancy so that we can recover from errors when we transmit our digital voice over the radio channel. Channel coding add s redun da ncy bi ts to the original information to detect and correct, errors occurred during transmission. GSM uses convol uti on coding and int erlea ving to achieve this protection. The exact algorithms used differ for speech and for different data rates
Channel Coding
In digital transmission, the quality of the transmitted signal is often expressed in terms of how many of the received bits are incorrect. This is called Bit Error Rate (BER). BER defines the percentage of the total number of received bits which are incorrectly detected.
This percentage should be as low as possible. It is not possible to reduce the percentage to zero because the transmission path is constantly changing.
Channel coding is used to detect and correct errors in a received bit stream. It adds bits to a message. These bits enable a channel decoder to determine whether the message has faulty bits, and to potentially correct the faulty bits.
Channel coding for GSM speech
Recall that the RPE-LTP Encoder produces a block of 260 bits every 20 ms. It was found (though testing) that some of the 260 bits were more important when compared to others. Below is the composition of these 260 bits. Cl as s Ia - 50 bits (most sensitive to bit errors) Cl as s Ib - 132 bits (moderately sensitive to bit errors) Cl as s II - 78 bits (least sensitive to error)
As a result of some bits being more important than others, GSM adds redundancy bits to each of the three Classes differently. The Class IA bits are encoded in a cyclic encoder. The Class Ib bits (together with the encoded Class IA bits) are encoded using convolutional encoding. Finally, the Class II bits are merely added to the result of the convolutional encoder.
Class Ia bits have a 3 bit Cyclic Redundancy Code added for error detection. 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 ½ rate convolutional encoder. Each input bit is encoded as two output bits. The convolutional encoder thus outputs 378 bits, which are added to the 78 remaining Class II bits, which are unprotected. Thus every 20 m sec speech sample is encoded as 456 bits, giving a bit rate of 22.8 kbps
Interleaving
To further protect against the burst errors common to the radio interface, each sample is interleaved. This method rea rra nge s a group of bits in a particular way. After encoding resultant sample block consists of 456 bits. These blocks are then divided into eight blocks each containing 57 bits. The first four blocks will be placed in the even bit positions of the first four bursts. The last four blocks will be placed in the odd bit positions of the next four bursts.
Because of interleaving lost bits are part of several different packets and each packet loses only a few bits out of a large number of bits. So Interleaving decreases the possibility of losing whol e bur st s during the transmission, by di spers in g the errors. Since the errors become less concentrated, it is then easier to correct them.
Encryp ti on
It is used to protect signaling and data. This process is done using A3, A5 and A8 algorithms
Modula tion The modulation chosen for the GSM system is the Ga us si an M ini mum Shi ft K eyin g (GMSK ) .
Discontinuous Transmission (DTX)
Discontinuous Transmission (DTX) is a method of saving battery power for the MS. An MS with the DTX function detects the input "voice" and turns the transmitter ON only while "voice“ is present. When there is no voice input, the transmitter is turned OFF.
Discontinuous transmission (DTX)
So DTX is used to suspend the radio transmission during the si lence periods. This exploits the observation that only 4050% during a conversation does the speaker actually talk. DTX helps also to reduce interference between different cells and to increase system capacity. An added benefit of DTX is that power is conserved at the mobile unit.
Voic e A ctivi ty D etection (VAD)
The DTX function is performed by means of VAD It is this which has to determine whether the sound represents speech or noise, even if the background noise is very important. If the voice signal is considered as noise, the transmitter is turned off producing then, an unpleasant effect called clipping.
Co mfort n oise
A side-effect of the DTX function is that when the signal is considered as noise, the transmitter is turned off and therefore, a total silence is heard at the receiver. This can be very annoying to the receiving user since it appears as a dead connection. In order to overcome this problem, the receiver creates a minimum of background noise called comfort noise. Comfort noise eliminates the impression that the connection is dead.
Pow er con trol
To minimize co-channel interference and to conserve power, both the mobiles and the Base Transceiver Stations operate at the lowest power level that will maintain an acceptable signal quality. The BTSs perform timing measurements; they also perform measurements on the power level of the different mobile stations. These power levels are adjusted so that the power is nearly the same for each burst. The BTS controls its power level. The MS measures the strength and the quality of the signal between itself and the BTS. If the mobile station does not receive correctly the signal, the BTS changes its power level and retransmits.
Dis con tin uou s r ecep tion
Another method used to conserve power at the MS is Discontinuous Reception (DRX). The paging channel, used by the BTS to signal an incoming call, is structured into subchannels. Each MS is assigned one of these subchannels and needs to listen only to its own sub-channel. In the time between successive paging subchannels, the mobile can go into “sleep mode”, when almost no power is used.
Timing Advance
In the GSM cellular mobile phone standard, timi ng ad vance value corresponds to the length of time a signal from the mobile phone takes to reach the base station. GSM uses TDMA technology in the radio interface to share a single frequency between several users, assigning sequential timeslots to the individual users sharing a frequency. Each user transmits periodically one-eighth of the time within one of the eight timeslots. Since the users are various distances from the base station and radio waves travel at the finite speed of light, the precise time at which the phone is allowed to transmit a burst of traffic within a timeslot must be adjusted accordingly. Timing Advance (TA) is the variable controlling this adjustment.
This synchronization between BTS and MS is achieved by using the concept of Timing Advance (TA). From the measurements, the BTS can calculate the Timing-Advance and send it back to the MS in the first downlink transmission. From the TA value received from the BTS, the MS know when to send the frame, so that it can arrive at the BTS in synchronism. The values of the TA is continuously calculated and transmitted to the MS during the call.
TRANSMISSION RATE
The amount of information transmitted over a radio channel over a period of time is known as the transmission rate. Transmission rate is expressed in bits per second or bit/s. In GSM the net bit rate over the air interface is 270kbit/s.
Channel Coding
The following figure shows the steps involved to transform speech audio to radio waves and vice versa.
GSM Speech Processing Steps
Speech compressed using a predictive coding scheme Divided into blocks, each of which is protected partly by cyclic code and partly by a convolutional code Interleaving to protect against burst errors Encryption for providing privacy Assembled into time slots Modulated for analog transmission using GMSK
Speech coding
The GSM speech codec that transforms the analog signal (voice) into a digital representation, must meet the following criteria: Maintain speech quality. Reduce redundancy in voice utterances. This reduction is essential due to transmission capacity limitation on the data channel. Adopt low complexity speech codec to reduce production costs.
GSM TRANSMISSION PROCESS
STAGE 1: ANALOG TO DIGITAL (A/D) CONVERSION STAGE 2: SEGMENTATION STAGE 3: SPEECH CODING STAGE 4: CHANNEL CODING STAGE 5: INTERLEAVING STAGE 6: CIPHERING/ENCRYPTION STAGE 7: BURST FORMATTING STAGE 8: MODULATION & TRANSMISSION
Speech Coding
In order to send the voice information across a radio network, first thing to be done is to turn the voice into a digital signal. GSM uses a method called RPE-LTP (Regular Pulse Excited - Long Term Prediction) with Linear Predictive Coding to turn our analog voice into a compressed digital equivalent. One of the primary functions of an MS is to convert the analog speech information into digital form for transmission using a digital signal. The analog to digital (A/D) conversion process outputs a collection of bits: binary ones and zeros which represent the speech input.
In modern phone systems, digital coding is used. The electrical variations induced into the microphone are sampled and each sample is then converted into a digital code. The voice waveform sampled at a rate of 8 kHz and sample is converted into an 8 bit binary number, representing 256 distinct values . Since we sample 8000 times per second and each sample is 8 binary bits, we have a bitrate of 8kHz X 8 bits = 64kbps.
This bitrate is unrealistic to transmit across a radio network. GSM speech coding works to compress the speech waveform into a sample that results in a lower bitrate using RPE-LTP. The speech signal is divided into blocks of 20ms . Once we have a digital signal we have to add some sort of redundancy so that we can recover from errors when we transmit our digital voice over the radio channel.
These blocks are then passed to the speech codec of 13 kbps, to obtain speech frames of 260 bits each.
GSM Channel Coding
Once we have a digital signal we have to add some sort of redundancy so that we can recover from errors when we transmit our digital voice over the radio channel. Channel coding add s redun da ncy bi ts to the original information to detect and correct, errors occurred during transmission. GSM uses convol uti on coding and int erlea ving to achieve this protection. The exact algorithms used differ for speech and for different data rates
Channel Coding
In digital transmission, the quality of the transmitted signal is often expressed in terms of how many of the received bits are incorrect. This is called Bit Error Rate (BER). BER defines the percentage of the total number of received bits which are incorrectly detected.
This percentage should be as low as possible. It is not possible to reduce the percentage to zero because the transmission path is constantly changing.
Channel coding is used to detect and correct errors in a received bit stream. It adds bits to a message. These bits enable a channel decoder to determine whether the message has faulty bits, and to potentially correct the faulty bits.
Channel coding for GSM speech
Recall that the RPE-LTP Encoder produces a block of 260 bits every 20 ms. It was found (though testing) that some of the 260 bits were more important when compared to others. Below is the composition of these 260 bits. Cl as s Ia - 50 bits (most sensitive to bit errors) Cl as s Ib - 132 bits (moderately sensitive to bit errors) Cl as s II - 78 bits (least sensitive to error)
As a result of some bits being more important than others, GSM adds redundancy bits to each of the three Classes differently. The Class IA bits are encoded in a cyclic encoder. The Class Ib bits (together with the encoded Class IA bits) are encoded using convolutional encoding. Finally, the Class II bits are merely added to the result of the convolutional encoder.
Class Ia bits have a 3 bit Cyclic Redundancy Code added for error detection. 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 ½ rate convolutional encoder. Each input bit is encoded as two output bits. The convolutional encoder thus outputs 378 bits, which are added to the 78 remaining Class II bits, which are unprotected. Thus every 20 m sec speech sample is encoded as 456 bits, giving a bit rate of 22.8 kbps
Interleaving
To further protect against the burst errors common to the radio interface, each sample is interleaved. This method rea rra nge s a group of bits in a particular way. After encoding resultant sample block consists of 456 bits. These blocks are then divided into eight blocks each containing 57 bits. The first four blocks will be placed in the even bit positions of the first four bursts. The last four blocks will be placed in the odd bit positions of the next four bursts.
Because of interleaving lost bits are part of several different packets and each packet loses only a few bits out of a large number of bits. So Interleaving decreases the possibility of losing whol e bur st s during the transmission, by di spers in g the errors. Since the errors become less concentrated, it is then easier to correct them.
Encryp ti on
It is used to protect signaling and data. This process is done using A3, A5 and A8 algorithms
Modula tion The modulation chosen for the GSM system is the Ga us si an M ini mum Shi ft K eyin g (GMSK ) .
Discontinuous Transmission (DTX)
Discontinuous Transmission (DTX) is a method of saving battery power for the MS. An MS with the DTX function detects the input "voice" and turns the transmitter ON only while "voice“ is present. When there is no voice input, the transmitter is turned OFF.
Discontinuous transmission (DTX)
So DTX is used to suspend the radio transmission during the si lence periods. This exploits the observation that only 4050% during a conversation does the speaker actually talk. DTX helps also to reduce interference between different cells and to increase system capacity. An added benefit of DTX is that power is conserved at the mobile unit.
Voic e A ctivi ty D etection (VAD)
The DTX function is performed by means of VAD It is this which has to determine whether the sound represents speech or noise, even if the background noise is very important. If the voice signal is considered as noise, the transmitter is turned off producing then, an unpleasant effect called clipping.
Co mfort n oise
A side-effect of the DTX function is that when the signal is considered as noise, the transmitter is turned off and therefore, a total silence is heard at the receiver. This can be very annoying to the receiving user since it appears as a dead connection. In order to overcome this problem, the receiver creates a minimum of background noise called comfort noise. Comfort noise eliminates the impression that the connection is dead.
Pow er con trol
To minimize co-channel interference and to conserve power, both the mobiles and the Base Transceiver Stations operate at the lowest power level that will maintain an acceptable signal quality. The BTSs perform timing measurements; they also perform measurements on the power level of the different mobile stations. These power levels are adjusted so that the power is nearly the same for each burst. The BTS controls its power level. The MS measures the strength and the quality of the signal between itself and the BTS. If the mobile station does not receive correctly the signal, the BTS changes its power level and retransmits.
Dis con tin uou s r ecep tion
Another method used to conserve power at the MS is Discontinuous Reception (DRX). The paging channel, used by the BTS to signal an incoming call, is structured into subchannels. Each MS is assigned one of these subchannels and needs to listen only to its own sub-channel. In the time between successive paging subchannels, the mobile can go into “sleep mode”, when almost no power is used.
Timing Advance
In the GSM cellular mobile phone standard, timi ng ad vance value corresponds to the length of time a signal from the mobile phone takes to reach the base station. GSM uses TDMA technology in the radio interface to share a single frequency between several users, assigning sequential timeslots to the individual users sharing a frequency. Each user transmits periodically one-eighth of the time within one of the eight timeslots. Since the users are various distances from the base station and radio waves travel at the finite speed of light, the precise time at which the phone is allowed to transmit a burst of traffic within a timeslot must be adjusted accordingly. Timing Advance (TA) is the variable controlling this adjustment.
This synchronization between BTS and MS is achieved by using the concept of Timing Advance (TA). From the measurements, the BTS can calculate the Timing-Advance and send it back to the MS in the first downlink transmission. From the TA value received from the BTS, the MS know when to send the frame, so that it can arrive at the BTS in synchronism. The values of the TA is continuously calculated and transmitted to the MS during the call.
TRANSMISSION RATE
The amount of information transmitted over a radio channel over a period of time is known as the transmission rate. Transmission rate is expressed in bits per second or bit/s. In GSM the net bit rate over the air interface is 270kbit/s.
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