Transmission and Receiving Chain

In the last few posts i wrote about the Speech Transmission in GSM, later i was reading more on it then i found this brief from a book. Flow Diagram below shows the transmitting and receiving chain of a GSM receiver. Several successive operations have to be performed to convert a speech signal into a radio signal and back. 

The following operations take place on the transmitting side:

• Source coding: Converts the analogue speech signal into a digital equivalent.
• Channel coding: Adds extra bits to the data flow. This way redundancy is introduced into the data flow, increasing its rate by adding information calculated from the source data, in order to allow detection or even correction of bit errors that might be introduced during transmission. This is described in more detail below.
• Interleaving: Consists of mixing up the bits of the coded data blocks. The goal is to have adjacent bits in the modulated signal spread out over several data blocks. The error probability of successive bits in the modulated stream is typically highly correlated, and the channel coding performance is better when errors are decorrelated. Therefore, interleaving improves the coding performance by decorrelating errors and their position in the coded blocks.
• Ciphering: Modifies the contents of these blocks through a secret code known only by the mobile station and the base station.
• Burst formatting: Adds synchronisation and equalisation information to the ciphered data. Part of this is the addition of a training sequence.
• Modulation: Transforms the binary signal into an analogue signal at the right frequency. Thereby the signal can be transmitted as radio waves.

The receiver side performs the reverse operations as follows:

• Demodulation: Transforms the radio signal received at the antenna into a binary signal. Today most demodulators also deliver an estimated probability of correctness for each bit. This extra information is referred to as soft decision or soft information.
• Deciphering: Modifies the bits by reversing the ciphering code.
• Deinterleaving: Puts the bits of the different bursts back in order to rebuild the original code words.
• Channel decoding: Tries to reconstruct the source information from the output of the demodulator, using the added coding bits to detect or correct possible errors, caused between the coding and the decoding.
• Source decoding: Converts the digitally decoded source information into an analogue signal to produce the speech.

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Satya Sravan

Ciphering & Modulation - Speech Transmission in GSM

Ciphering 

Ciphering is used to protect signaling and user data from intruder. First of all, a ciphering key (Kc) is computed using the algorithm A8 stored on the SIM card, the subscriber key (Ki, also stored in the SIM card) and a random number (RAND) delivered by the network (this random number is the same as the one used for the authentication procedure). Secondly, a 114 bit sequence is produced using the ciphering key, an algorithm called A5 and the TDMA frame number (Fn) number provided by the network. This bit sequence is then XORed (TRUE if only one of the inputs is TRUE) with the two 57 bit blocks of data included in a normal burst.

Modulation

The burst formatted bits are then modulated using GMSK which is nothing but 8PSK modulation technique and transmitted over a carrier. In GMSK the bit stream is first filtered using a Gaussian filter so as to remove the higher harmonics and round off the corners of the bit pulses. Then they are used for modulating the frequency of the carrier using MSK modulation.

That concludes one side of transmission; the decoding part happens in the other side. This is all together how speech transmission occurs in GSM

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Satya Sravan 

Burst Formatting - Speech Transmission in GSM

The GSM burst, or transmission can fulfill a variety of functions. Some GSM bursts are used for carrying data while others are used for control information.

Bursts and Frames

The information contained in one time slot on the TDMA frame is called a burst. The bit rate over the air interface is 270.8 kbps. This gives a bit time of 3.692 ms (48/13 ms). The time interval of TS thus corresponds to 156.25 bits. The physical content of TS is called a burst.

There are five different types of bursts:

• Normal Burst (NB): used to carry information on traffic and control channels.

• Frequency Correction Burst (FB): used for frequency synchronization of the mobile.

• Synchronization Burst (SB): used for frame synchronization of the mobile.

• Access Burst (AB): used for random access and handover access.

• Dummy Burst: used when no other type of burst is to be sent.

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Satya Sravan

Interleaving - Speech Transmission in GSM

Interleaving is a process of dispersing the bits of a data burst over multiple bursts in a systematic way. Benefit of this technique: when a data-burst is lost (due to burst error in the radio interface) it does not mean a 100% loss of a single burst rather a partial loss of many bursts

First level of interleaving

The channel coder provides 456 bits for every 20 ms of speech which are interleaved in eight blocks of 57 bits shown below.

In a normal burst, there is space for two of these speech blocks (Figure). Thus, if one burst transmission is lost, there is a 25% BER for the entire 20 ms of speech (2/8= 25%).

Second level of interleaving

If only one level of interleaving is used, a loss of this burst results in a total loss of 25%. This is too much for the channel decoder to correct. A second level of interleaving can be introduced to further reduce the possible BER to 12.5%.

Instead of sending two blocks of 57 bits from the same 20 ms of speech within one burst, a block from one 20 ms and a block from next sample of 20 ms are sent together. A delay is introduced in the system when the MS must wait for the next 20 ms of speech. However, the system can now afford to lose a whole burst, out of eight, as the loss is only 12.5% of the total bits from each 20ms speech frame. 12.5% is the maximum loss level that channel decoder can correct.

Speech Frame

The bits must then be sent over the air using a carrier frequency. GSM uses the GMSK modulation technique. The bits are modulated onto a carrier frequency and transmitted.

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Satya Sravan

Channel Coding - Speech Transmission in GSM

Channel coding in GSM uses the 260 bits from speech coding as input to channel coding and outputs 456 encoded bits.

The 260 bits are split according to their relative importance:

• Block 1: 50 very important bits
• Block 2: 132 important bits and
• Block 3: 78 not so important bits

The first block of 50 bits is sent through a block coder, which adds three parity bits that will result in 53 bits. These three bits are used to detect errors in a received message.

The 53 bits from first block, the 132 bits from the second block and 4 tail bits (total = 189) are sent to a 1:2 convolutional coder which outputs 378 bits. Bits added by the convolutional coder enable the correction of errors when the message is received.

The bits of block 3 are not protected.

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Satya Sravan

Speech Coding & Segmentation - Speech Transmission in GSM

In GSM, the speech coding process analyzes speech samples and outputs parameters of what the speech consists of the tone, length of tone, pitch, etc. This is then transmitted through the network to another MS, which generates the speech based on these parameters.

The process of segmentation and speech coding is explained in more detail as follows:

The human speech process starts in the vocal chords or speech organs, where a tone is generated. The mouth, tongue, teeth, etc. act as a filter, changing the nature of this tone. The aim of speech coding in GSM is to send only information about the original tone itself and about the filter.

Segmentation:- Given that the speech organs are relatively slow in adapting to changes, the filter parameters representing the speech organs are approximately constant during 20 ms. For this reason, when coding speech in GSM, a block of 20 ms is coded into one set of bits. In effect, it is similar to sampling speech at a rate of 50 times per second instead of the 8,000 used by A/D conversion.

Speech Coding:- Instead of using 13 bits per sample as in A/D conversion, GSM speech coding uses 260 bits. This calculates as 50 x 260 = 13 Kbits/s. This provides a speech quality which is acceptable for mobile telephony and comparable with wire-line PSTN phones.

Many types of speech coders are available. Some offer better speech quality, at the expense of a higher bit rate (waveform coders). Others use lower bit rates, at the expense of lower speech quality (vocoders). The hybrid coder which GSM uses provides good speech quality with a relatively low bit rate, at the expense of speech coder complexity.

The GSM speech coder produces a bit rate of 13 kbits/s per subscriber. When it is considered that 8 subscribers use one radio channel, the overall bit rate would be 8 x 13 kbits/s = 104 kbits/s. This compares favorably with the 832 Kbits/s from A/D conversion.

However, speech coding does not consider the problems which may be encountered on the radio transmission path. The next stages in the transmission process, channel coding and interleaving, help to overcome these problems.

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Satya Sravan

Speech Transmission in GSM [Intro]

In the series of posts, i wish to write about how the actual speech that you talk gets transmitted at the other end. The whole process is shown in this illustration along with the rate at which the transfer occurs. 

The steps involved at one end are 

• Speed Coding and Segmentation
• Channel Coding
• Interleaving
• Burst Formatting 
• Ciphering 
• Modulation

And at the end the counterpart is gonna happen. I will discuss each step in separate post. 

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Satya Sravan