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Function apps running under the Apps Service plan may have multiple slots, while under Consumption only one slot is allowed. The following reflect how functions are affected by swapping slots: Traffic redirection is seamless; no requests are dropped because of a swap.

GSM primer includes:
GSM introductionNetwork architectureNetwork interfacesRF interface / slot & burstGSM framesPower classes & controlChannelsAudio codecs / vocodersHandover

A variety of different forms of audio codec or vocoder are available for general use, and the GSM system supports a number of specific audio codecs. These include the RPE-LPC, half rate, and AMR codecs. The performance of each voice codec is different and they may be used under different conditions, although the AMR codec is now the most widely used. Also the newer AMR wideband (AMR-WB) codec is being introduced into many areas, including GSM

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Voice codec technology has advanced by considerable degrees in recent years as a result of the increasing processing power available. This has meant that the voice codecs used in the GSM system have large improvements since the first GSM phones were introduced.

Vocoder / codec basics

Vocoders or speech codecs are used within many areas of voice communications. Obviously the focus here is on GSM audio codecs or vocoders, but the same principles apply to any form of codec.

If speech were digitised in a linear fashion it would require a high data rate that would occupy a very wide bandwidth. As bandwidth is normally limited in any communications system, it is necessary to compress the data to send it through the available channel. Once through the channel it can then be expanded to regenerate the audio in a fashion that is as close to the original as possible.

To meet the requirements of the codec system, the speech must be captured at a high enough sample rate and resolution to allow clear reproduction of the original sound. It must then be compressed in such a way as to maintain the fidelity of the audio over a limited bit rate, error-prone wireless transmission channel.

Audio codecs or vocoders can use a variety of techniques, but many modern audio codecs use a technique known as linear prediction. In many ways this can be likened to a mathematical modelling of the human vocal tract. To achieve this the spectral envelope of the signal is estimated using a filter technique. Even where signals with many non-harmonically related signals are used it is possible for voice codecs to give very large levels of compression.

A variety of different codec methodologies are used for GSM codecs:

• CELP: The CELP or Code Excited Linear Prediction codec is a vocoder algorithm that was originally proposed in 1985 and gave a significant improvement over other voice codecs of the day. The basic principle of the CELP codec has been developed and used as the basis of other voice codecs including ACELP, RCELP, VSELP, etc. As such the CELP codec methodology is now the most widely used speech coding algorithm. Accordingly CELP is now used as a generic term for a particular class of vocoders or speech codecs and not a particular codec.
  The main principle behind the CELP codec is that is uses a principle known as 'Analysis by Synthesis'. In this process, the encoding is performed by perceptually optimising the decoded signal in a closed loop system. One way in which this could be achieved is to compare a variety of generated bit streams and choose the one that produces the best sounding signal.
• ACELP codec: The ACELP or Algebraic Code Excited Linear Prediction codec. The ACELP codec or vocoder algorithm is a development of the CELP model. However the ACELP codec codebooks have a specific algebraic structure as indicated by the name.
• VSELP codec: The VSELP or Vector Sum Excitation Linear Prediction codec. One of the major drawbacks of the VSELP codec is its limited ability to code non-speech sounds. This means that it performs poorly in the presence of noise. As a result this voice codec is not now as widely used, other newer speech codecs being preferred and offering far superior performance.

GSM audio codecs / vocoders

A variety of GSM audio codecs / vocoders are supported. These have been introduced at different times, and have different levels of performance. Although some of the early audio codecs are not as widely used these days, they are still described here as they form part of the GSM system.

GSM AMR Codec

The AMR, Adaptive Multi-rate codec is now the most widely used GSM codec. The AMR codec was adopted by 3GPP in October 1988 and it is used for both GSM and circuit switched UMTS / WCDMA voice calls.

The AMR codec provides a variety of options for one of eight different bit rates as described in the table below. The bit rates are based on frames that are 20 millisceonds long and contain 160 samples. The AMR codec uses a variety of different techniques to provide the data compression. The ACELP codec is used as the basis of the overall speech codec, but other techniques are used in addition to this. Discontinuous transmission is employed so that when there is no speech activity the transmission is cut. Additionally Voice Activity Detection (VAD) is used to indicate when there is only background noise and no speech. Additionally to provide the feedback for the user that the connection is still present, a Comfort Noise Generator (CNG) is used to provide some background noise, even when no speech data is being transmitted. This is added locally at the receiver.

The use of the AMR codec also requires that optimized link adaptation is used so that the optimum data rate is selected to meet the requirements of the current radio channel conditions including its signal to noise ratio and capacity. This is achieved by reducing the source coding and increasing the channel coding. Although there is a reduction in voice clarity, the network connection is more robust and the link is maintained without dropout. Improvement levels of between 4 and 6 dB may be experienced. However network operators are able to prioritise each station for either quality or capacity.

The AMR codec has a total of eight rates: eight are available at full rate (FR), while six are available at half rate (HR). This gives a total of fourteen different modes.

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AMR-WB codec

Adaptive Multi-Rate Wideband, AMR-WB codec, also known under its ITU designation of G.722.2, is based on the earlier popular Adaptive Multi-Rate, AMR codec. AMR-WB also uses an ACELP basis for its operation, but it has been further developed and AMR-WB provides improved speech quality as a result of the wider speech bandwidth that it encodes. AMR-WB has a bandwidth extending from 50 - 7000 Hz which is significantly wider than the 300 - 3400 Hz bandwidths used by standard telephones. However this comes at the cost of additional processing, but with advances in IC technology in recent years, this is perfectly acceptable.

The AMR-WB codec contains a number of functional areas: it primarily includes a set of fixed rate speech and channel codec modes. It also includes other codec functions including: a Voice Activity Detector (VAD); Discontinuous Transmission (DTX) functionality for GSM; and Source Controlled Rate (SCR) functionality for UMTS applications. Further functionality includes in-band signaling for codec mode transmission, and link adaptation for control of the mode selection.

The AMR-WB codec has a 16 kHz sampling rate and the coding is performed in blocks of 20 ms. There are two frequency bands that are used: 50-6400 Hz and 6400-7000 Hz. These are coded separately to reduce the codec complexity. This split also serves to focus the bit allocation into the subjectively most important frequency range.

The lower frequency band uses an ACELP codec algorithm, although a number of additional features have been included to improve the subjective quality of the audio. Linear prediction analysis is performed once per 20 ms frame. Also, fixed and adaptive excitation codebooks aresearched every 5 ms for optimal codec parameter values.

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The higher frequency band adds some of the naturalness and personality features to the voice. The audio is reconstructed using the parameters from the lower band as well as using random excitation. As the level of power in this band is less than that of the lower band, the gain is adjusted relative to the lower band, but based on voicing information. The signal content of the higher band is reconstructed by using an linear predictive filter which generates information from the lower band filter.

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Not all phones equipped with AMR-WB will be able to access all the data rates - the different functions on the phone may not require all to be active for example. As a result, it is necessary to inform the network about which rates are available and thereby simplify the negotiation between the handset and the network. To achieve this there are three difference AMR-WB configurations that are available:

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• Configuration A: 6.6, 8.85, and 12.65 kbit/s
• Configuration B: 6.6, 8.85, 12.65, and 15.85 kbit/s
• Configuration C: 6.6, 8.85, 12.65, and 23.85 kbit/s

Panaji india. It can be seen that only the 23.85, 15.85, 12.65, 8.85 and 6.60 kbit/s modes are used. Based on listening tests, it was considered that these five modes were sufficient for a high quality speech telephony service. The other data rates were retained and can be used for other purposes including multimedia messaging, streaming audio, etc.

Although these have been described as GSM codecs, they are also used in a number of other areas as well - some are used with the circuit switched voice employed in UMTS. The codec performance has improved since the first GSM codecs were introduced: speech quality along with the bandwidth required have all been improved.

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