GB2367209A - Communication terminal with voice signal buffering - Google Patents

Abstract

A communication unit (100) comprises a speech processing unit (130) operably coupled to a memory element (116) and a processor (108), wherein the processor (108) determines whether a communication link for the communication unit (100) has been established. The speech processing unit (130) receives a speech signal and determines whether to buffer in the memory element (116) a first portion of speech (404) for subsequent transmission, dependent upon this determination. This ensures that the opening portion of speech is not lost while the call is being set-up.

Description

COMMUNICATION SYSTEM, COMMUNICATION UNIT AND METHOD FOR PROVIDING A COMMUNICATION LINK Field of the Invention This invention relates to the provision of a voice communication link in a voice communication system. The invention is applicable to, but not limited to, the provision of a pre-call set-up arrangement in a wireless voice communication system.

Background of the Invention In the field of this invention it is known that wireless communications systems, for example cellular telephony or private mobile radio (PMR) communications systems, typically provide for radio telecommunication links to be arranged between a plurality of base transceiver stations (BTSs) and a plurality of subscriber units, often termed mobile stations (MSs).

In the art, the term mobile station generally includes both hand-portable and vehicular mounted radio units.

Furthermore, the communications link from a BTS to a MS is referred as to the down-link path. Conversely, the communications link from a MS to a BTS is referred to as the up-link path.

A popular version of the PMR communication system is twoway radio, where MSs communicate directly to each other, without the need for the communication to be processed through BTSs. A repeater station is often provided to enhance a basic two-way communication, typically in order to extend the range of the communication link. The repeater station generally performs minimal processing of received signals, typically sufficient processing to correct for any errors introduced in the up-link transmission, before re-transmitting (repeating) the received transmission to the intended recipient in the down-link path.

A wireless communication system is distinguished over a fixed communication system, such as a public switched telephone network (PSTN), principally in that MSs are mobile (in the general sense) and therefore may move between service providers (and/or different BTS). In doing so the MSs encounter varying radio propagation environments.

In a two way radio communication system there are often a number of communication resources, for example frequencies, time periods, coding schemes, that are available for allocating to communication system users to enable the users to set up a communication link. In some communication systems, the radio users transmit and receive on the same radio frequency. In such a system, assuming no geographical reuse of the frequency, only one call can be established between at least two radio units

at any one time. It is possible, and often implemented in multi-site systems, that the same communication resources are re-allocated on a geographical basis to some users. This re-allocation of resource procedure is followed as long as the communication link would not interfere with existing users using that particular resource, for example where any potential interfering signal level from new users would be too low to be detected by the existing users of that resource.

In two-way radio communications systems, communication between two radios is typically performed via individual calls between two users or via group calls where users communicate to a number of other users in a particular communication group. In mobile trunked radio communication systems, it is known to record (or log) voice communications at a central location of the system, for example a central controller or primary base transceiver station. A central location is the optimal position for performing voice recording as all voice communications typically pass through such a position in a trunked radio communication system.

An example of the necessity for logging voice communications in such a central location is found in Public Safety Systems where a voice record of say, a police operation is logged. The information stored in such a process typically includes, in addition to the record of actual voice communication, information

relating to the date, time, length of time of the active communication link, who was involved, etc.

One communications standard that is presently being defined by the European Telecommunications Standards Institute (ETSI) is the Digital Interchange of Information and Signalling standard (DIIS). The focus of the DIIS standard is the transition from the analogue technology in today's low-tier Professional Mobile Radio systems to a higher speed (12 kbits/sec in a 12.5kHz and above frequency channel spacing) digital technology supporting speech and data.

The type of communications offered by DIIS technology include: (i) MS to MS communications (where there is no infrastructure, limited site coverage and single frequency operation is the accepted standard practice) for which simplex or semi-duplex communications are generally provided; (ii) BTS to MS (and conversely MS to BTS) communications, for example a dispatch function for fleet, taxi services, for which semi-duplex communications are generally provided; (iii) repeater mode of operation (where the communication range is extended by use of a remotely sited repeater or BTS); and (iv) networked repeater mode of operation (whereby multiple site coverage is provided by relayed repeaters or simulcast systems).

In the particular context of a Public Safety System, one of the major problems associated with such systems is the call set-up time. The typical explanation used to describe the well-known call set-up problem is with the "shoot"versus"don't shoot"scenario. In emergency calls, the user wants to be able to have security in a call, with the call commencing immediately a push-to-talk button is depressed.

However, with normal call set-up times of a few hundred milli-seconds, the initial part of a voice communication can be lost, whilst the call is being set-up. The worst scenario, often used to highlight the importance of fast call set-up times when designing or selling communication systems, is when the"don't"portion of an initial speech signal is lost, leaving"shoot"as the first utterance of speech, once a communication link has been established.

In order to address this problem, manufacturers and standards bodies focus on minimizing call set-up times.

However, given the need to perform synchronisation, registration, channel access request, request acknowledgement and then allocation of a communication channel, there is a limit to how much a reduction in call set-up times can address the aforementioned problem.

Thus, there exists a need in the field of this invention to provide a communications system and communications units that can initiate a call set-up wherein the

abovementioned truncated speech disadvantage may be alleviated.

Summary of the Invention In accordance with a first aspect of the preferred embodiment of the present invention, a communication unit in accordance with claim 1 is provided.

In a second aspect of the preferred embodiment of the present invention, a simplex communication system, semiduplex communication system, radio communication system or fixed communication system, having at least one processing element adapted to accommodate the speech buffering and subsequent transmission of speech signals from a communication unit in accordance with any one of claims 1 to 9 is provided.

In a third aspect of the preferred embodiment of the present invention, a method for providing a communication link to a communication unit in a communication system, in accordance with claim 11 is provided.

Exemplary embodiments of the present invention will now be described, with reference to the accompanying drawings, in which:

FIG. 1 shows a block diagram of a communications unit that can be adapted to support the various inventive concepts of a preferred embodiment of the present invention; FIG. 2 shows a block diagram of a preferred vocoder arrangement adapted to support the various inventive concepts of a preferred embodiment of the present invention; FIG. 3 shows a speech transmission system in accordance with a preferred embodiment of the invention ; FIG. 4 (a)- (c) show a series of speech input signals exemplifying the benefits associated with the inventive concepts of the preferred embodiment of the invention ; and FIG. 5 shows a flowchart of a method of providing a communication link, whereby truncation of a speech signal is avoided, in accordance with a preferred embodiment of the invention.

Description of Preferred Embodiment (s) Turning now to FIG. 1, there is shown a block diagram of a MS 100 adapted to support the inventive concepts of the preferred embodiments of the present invention. The MS 100 contains an antenna 102 preferably coupled to a duplex filter or circulator 104 that provides isolation between a receiver and a transmitter chain within the MS 100.

As known in the art, the receiver chain typically includes scanning receiver front-end circuitry 106 (effectively providing reception, filtering and intermediate or base-band frequency conversion). The scanning front-end circuit is serially-coupled to a signal processing function 108. An output from the signal processing function is provided to a suitable output device 110, such as a speaker via a speech processing unit 130.

The speech processing unit 130 includes a speech encoding function 134 to encode a user's speech signals into a format suitable for transmitting over the transmission medium. The speech processing unit 130 also includes a speech decoding function 132 to decode received speech signals into a format suitable for outputting via the output device (speaker) 110. The speech processing unit 130 is operably coupled to a memory unit 116, via link 136, and a timer 118 via a controller 114. In particular, the operation of the speech processing unit

130 has been adapted to support the inventive concepts of the preferred embodiments of the present invention.

The receiver chain also includes received signal strength indicator (RSSI) circuitry 112 (shown coupled to the scanning receiver front-end 106, although the RSSI circuitry 112 could be located elsewhere within the receiver chain). The RSSI circuitry is coupled to a controller 114 for maintaining overall subscriber unit control. The controller 114 is also coupled to the scanning receiver front-end circuitry 106 and the signal processing function 108 (generally realised by a DSP).

The controller 114 may therefore receive bit error rate (BER) or frame error rate (FER) data from recovered information. The controller 114 is coupled to the memory device 116 for storing operating regimes, such as decoding/encoding functions and the like. A timer 118 is typically coupled to the controller 114 to control the timing of operations (transmission or reception of timedependent signals) within the MS 100.

In the context of the present invention, the timer 118 controls the timing of speech signals, in the transmit (encoding) path and/or the receive (decoding) path, to ensure that all speech is transmitted, or received, irrespective of whether such speech signals were initiated before a communication link had been set up.

As regards the transmit chain, this essentially includes an input device 120, such as a microphone transducer coupled in series via speech encoder 134 to a transmitter/modulation circuit 122. Thereafter, any transmit signal is passed through a power amplifier 124 to be radiated from the antenna 102. The transmitter/modulation circuitry 122 and the power amplifier 124 are operationally responsive to the controller, with an output from the power amplifier coupled to the duplex filter or circulator 104. The transmitter/modulation circuitry 122 and scanning receiver front-end circuitry 106 comprise frequency upconversion and frequency down-conversion functions (not shown).

In operation, if the MS 100 is initiating a communication link, the speech encoder receives an input speech signal, preferably once a push-to-talk button has been depressed.

The controller 114, operably coupled to the speech encoder 134, determines whether a communication link has been established using, for example, a standard call setup procedure. If a communication link has not yet been established, the controller 114 directs the initial 'transmit'speech segments to into buffers (not shown) within memory function 116, under control of the timer 118. Once a communication link has been established, the controller directs the initial'transmit'speech segments from the buffers, in the order that they were stored, back into the transmit path to transmitter/modulator circuitry 122.

In the preferred embodiment of the invention, these initial'transmit'speech segments are processed at a slightly faster speed than the'true'speech input to microphone 120. In the meantime, subsequent'transmit' speech segments are also processed into buffers within the memory function 116 at the normal'true'speech rate. This action, of storing input speech at the'true'speech rate whilst subsequently processing and transmitting speech at a'faster'speech rate, is performed until the buffers are empty.

The emptying of the buffers indicates that the'faster' speech being processed and transmitted, has caught up with the'true'speech being input to the microphone 120.

When this happens, the normal speech path of microphone 120 to speech encoder 134 to transmitter/modulation circuitry 122 is resumed.

In the converse operational mode, if the MS 100 is receiving signals from a transmitting unit employing the aforementioned store and process technique, the communication link has clearly already been established.

The receiver's signal processing function 108 (or the speech decoder function 132) receives an input speech signal, and determines that the received speech signal has been transmitted at the'faster'speech rate. The controller 114 controls the processing and decoding of

these initial'received'speech segments at the'faster' speech rate.

Dependent upon the rate of this'faster'speech rate, the controller may decide to buffer the received speech signals in the memory unit 116, if the rate is deemed too fast for useful perceptual hearing by a user. In such a case, the action of storing received speech at the 'faster', and subsequently'true', speech rate, whilst processing the received speech at an intermediate rate until the received speech catches up with the buffered speech, is performed until the buffers are empty.

The emptying of the buffers indicates that the'faster' speech being initially processed and received, has caught

up with the'true'speech that is subsequently being received, that is input to the speaker 110. When this happens, the normal speech path of receiver 106 to signal processor 108 to speech decoder 132 and thereby to speaker 110 is resumed.

Of course, the various components within the MS 100 can be arranged in any suitable functional topology able to utilise the inventive concepts of the present invention. Furthermore, the various components within the MS 100 can be realised in discrete or integrated component form, with an ultimate structure therefore being merely an arbitrary selection.

It is within the contemplation of the invention that the preferred use of speech storing can be implemented in software, firmware or hardware, with the function being

implemented in a software processor (or indeed a digital signal processor (DSP)), performing the speech processing function, merely a preferred option.

Turning now to FIG. 2, the preferred embodiment of the invention is described with reference to a dual-rate code excited linear predictive (CELP) speech processor, as recommended in the DIIS standard. Clearly, a dual-rate codec is a sub-set of a multi-rate codec arrangement that could equally benefit from the inventive concepts described herein.

FIG. 2 shows a block diagram of a preferred vocoder 200, adapted to support the various inventive concepts of a preferred embodiment of the present invention. An acoustic input signal is applied to the speech vocoder 200 at microphone 202. The input signal is converted into a sequence of N pulse samples, and the amplitude of each pulse sample is then represented in a digital code by analog-to-digital (A/D) converter 108, as known in the art.

The digital output of A/D 108, which may be represented as input speech vector s (n), is then applied to coefficient analyser 210. For each block of speech, a set of linear predictive coding (LPC) parameters is produced. The computed LPC parameters for the input speech signal are then quantized in LPC quantizer 212.

LPC quantizer selects, via switch 234, either of two LPC codebooks 232,233, to quantize the input speech signal.

In the preferred embodiment of the invention, the two LPC codebooks 232, 233 represent respective CELP codebooks in a dual CELP codebook arrangement. The selection of the appropriate LPC codebook (232 or 233) is selected using a speech input rate control signal 204.

In accordance with a preferred embodiment of the invention, should the controller 114 of MS 100 determine that a communication link has not yet been established, the controller 114 buffers (not shown in FIG. 2) the input speech signal until the communication link has been established. Once the communication link is established, the controller 114 controls the speech input rate control signal 204 to select the LPC codebook at the faster codec rate, irrespective of whether the initial speech was at this faster rate.

In the preferred embodiment of the invention, if the "true"speech input is at the slower rate, the'transmit' speech segments are quantized and processed at the faster rate until such time as the quantized faster rate signals catch up with the slower rate signals input to the speech encoder. At this time, the controller 114 switches the LPC codebook to the slower rate codebook via the speech input rate control signal 204.

If the input speech signal was employed at the faster rate, alternative or complementary mechanisms, such as that described with reference to FIG. 3 or FIG. 4, can be

employed to enable the quantized speech signal to"catch up"with the"true"input speech signal.

The generated, quantized LPC parameters are applied to multiplexer 250 for sending over the communications channel 252 for use by a speech synthesizer/decoder at the receiving unit.

The generated, quantized LPC parameters are also applied to a LPC synthesis filter 224, that forms part of an adaptive feedback loop in generating excitation parameters that best represent the input speech signals.

The LPC synthesis filter output is subtracted from the input speech vector s (n) at subtractor 230, as shown.

A codebook search controller 240 selects the best indices and gains from either of the two fixed codebooks 214,215 that represent the dual rate speech operation and an adaptive codebook 216, in order to produce a minimum weighted error in the summed chosen excitation vector used to represent the input speech sample.

The selection of the fixed codebook (214 or 215) is again determined by a switch 226, under the control of controller 114, via the speech input rate control signal 204. The output of the selected fixed codebook (say fixed codebook 214 as shown) and the second adaptive codebook 216 are input into respective gain functions 222 and 218. The gain-adjusted outputs are then summed in summer 220 and input into the LPC synthesis filter 224.

The output from the summer 220 is fed back to the adaptive codebook 216 in a feedback loop, as known in the art.

A more detailed description of the functionality of a typical speech encoding unit can be found in"Digital speech coding for low-bit rate communications systems"by A. M. Kondoz, published by John Wiley in 1994.

The inventors of the present invention have recognised the opportunity to use a dual rate codebook, or at least a full-rate and half-rate codebook as per the recommendations in the GSM standard, to provide this catch-up mechanism for quantizing and synthesizing stored speech signals.

In the preferred embodiment of the invention, the speech input rate control signal 204 is also transmitted to the receiving unit, over communications channel 252 and via multiplexer 250, in order for the decoder to recognise and adapt its decoding operation according to the rate that is applied to particular portions of the speech signal.

Although not shown, the decoder operation performs the converse receiving operation on the transmitted and received speech signal in order to adjust the received speech signal so that there is minimum disruption to the user.

The use of two LPC codebooks 232, 233 and first and second fixed codebooks 214,215 is only a preferred solution adapted to the DIIS dual rate vocoder implementation. However, it is within the contemplation of the invention that any number of LPC and speech codebooks would benefit from the inventive concepts described herein.

In DIIS, for the purpose of offering duplex operation, together with range extension, there is a dual rate vocoder. By using a dual rate vocoder, the preferred operation is to record the initial speech at the lower (range-extended) bit-rate. Once the link has been established, the communications unit will commence transmitting the lower-rate recorded speech at a higher bit-rate until the buffer is empty. Once the buffer is empty, the speech processing operation will revert to normal lower-rate operation.

At the receiving unit, when lower-rate recorded speech is received at a higher-rate, the speech is processed at slightly faster than'true'lower-rate whilst buffering up the extra speech, until the extra data has been processed. Subsequently, the speech processing reverts back to normal lower-rate operation.

Turning now to FIG. 3, an extension to a speech transmission system in accordance with another aspect of a preferred embodiment of the invention is shown. The

extension to a speech transmission system is described as an alternative"catch-up,"mechanism.

Input speech is input to microphone 302 and preferably passed to a silence compression unit 304 that recognises periods of silence and compresses the silence periods in speech in order to perceptually speed up the actual voice portions of the input speech signal. The silence compression unit 304 outputs the speech signal to speech encoder 306 for encoding the speech, for example in accordance with the speech encoder shown in FIG. 2. The silence compression unit 304 and speech encoder 306 are controlled in accordance with a speech encoder rate 316, as shown.

The output from the speech encoder 306 is input into a data buffer 308, for temporarily storing speech signals.

The output from the data buffer 308 is input into forward error correction (FEC) encoder 310. The FEC encoder outputs FEC encoded speech signals to a second data buffer 312, which then forwards the signals to the modulation and transmitter circuitry. The FEC encoder 310 is controlled in accordance with the FEC encoder rate 314, as shown. Data buffer 308 provides an input into the speech encoder rate 316 and together with second data buffer 312 provides an input into the FEC encoder rate 314 to determine the optimal rates for the speech buffering process, whilst a communication link is being established.

Advantageously, with such an arrangement as described with reference to FIG. 3, the opportunities for temporarily storing speech before transmission include: (i) compressing silence periods; (ii) controlling the speech encoder rate (as per FIG. 2); and/or (iii) controlling the FEC encoding rate.

It is within the contemplation of the invention that the aforementioned speech catch-up techniques can be used in isolation, or can be viewed as complementary to one another and, as such, more than one technique can be used at any particular instant.

Additionally, the determination of the speech encoder rate 316 and determination of the FEC encoder rate 314 are dependent upon the modulation rate 318. The modulation rate is determined by the data transmission rate 322, once a push to talk input 320 has been initiated.

It is within the contemplation of the invention that other receiver and speech processing circuit arrangements would benefit from the inventive concepts described herein. For example, a silence compressor function may be placed in either, or both, the speech encoder or speech decoder.

FIG's 4a-4c show an example of how a pulse coded modulation (PCM) technique can be adapted in accordance with a preferred embodiment of the invention.

FIG. 4a shows a typical input speech signal 400, with the audio amplitude shown on the vertical axis plotted against time on the horizontal axis. Timing points of interest on the input speech signal 400 include the moment that PTT is depressed 402, indicating the time when the user wishes to initiate a communication. As is usual in such radio communication systems, the user starts to talk as soon as PTT is depressed, and often before the call set-up has been completed, as identified earlier.

The time when the call set-up has been completed is shown at timing position 406, highlighting the fact that, ordinarily, speech portion 404 will have been lost to the receiving unit. Input speech is continued until the speech signal terminates at timing position 408, followed by the PTT being released by the user at timing position 410 and finally the communication link being disabled at timing position 412.

The conventional speech output 430 from the transmitting unit is shown in FIG. 4b. As indicated above, the initial portion of the speech 404 (in FIG. 4a) has been lost as the communication link has not yet been set up.

Once the communication link has been set-up at timing position 432, the speech communication continues until

timing position 438 indicating the termination of the speech signal. Points of interest include the periods of silence shown as speech gaps 434 and 436 respectively.

In accordance with a preferred embodiment of the present invention, namely the buffering of speech signals coupled to the use of silence compression, FIG. 4c shows the output speech signal 460 when incorporating at least some of the inventive concepts described herein.

The communication link is established at point 462. Note that there is an initial non-speech transmission period, due to processing by the speech encoder in recognising that a communication link is being established, before buffered speech is output. The initial portion of the transmitted speech 464 incorporates the speech 404 initially uttered by the user, and buffered in accordance with a preferred embodiment of the present invention.

The silence periods 466 and 468 have been compressed so that the transmitted speech signals can catch up with the input speech signals. This is shown in relation to realtime speech signal 470, where the buffering and subsequent silence compression portions 466,468 have enabled the speech encoder to catch up with the input speech signal. The remaining portion of the speech signal is encoded in real-time, until the speech communication is terminated at point 472.

The beneficial opportunity to use time (silence or PCM) compression could be implemented before vocoding is performed, as shown in FIG. 3. This results in a slight increase in delay in the initial part of the transmission. This portion of speech is replayed at the receiver in slightly faster than real time.

Advantageously, the opening portion of speech is not truncated.

Turning now to FIG. 5, a flowchart of a method of providing a communication link is shown, whereby truncation of a speech signal is avoided in accordance with a preferred embodiment of the invention.

Once a press-to-talk button (or similar) is depressed by the user, indicating a request for a communication channel as shown in step 500, the communication system will initiate a call set-up process. The user will commence uttering speech, as in step 502, and once speech has been determined by the communication unit, a determination is made as to whether a communication link has been established, as shown in step 504. If a communication link has been established, a normal speech communication can take place, as shown in step 520.

However, if a communication link has not been established in step 504, after the user has commenced uttering speech in step 502, the initial (and perhaps subsequent dependent upon the length of time required to catch-up) portions of speech are buffered in the communication

unit, as shown in step 506. The communications unit continues to determine whether a communication link has been established, as in step 508 If not, speech is continued to be buffered, as shown in step 506.

Once a communication link has been established in step 508, speech is processed using one or more of the aforementioned catch-up mechanism (s), as in step 510.

The catch-up mechanism (s) include: using a higher-rate speech coder than the true speech rate coder in a dual or multi-rate coder, use time (PCM and/or silence) compression, adapt the FEC encoder rate. The speech signal is then transmitted, as shown in step 512.

At the receiving unit, the received speech signal is processed in accordance with the selected catch-up mechanism (s), as shown in step 514. A determination is made, by both the transmitting unit and the receiving unit, as to whether the speech processing rate has caught up with the input speech rate, i. e. buffers are empty, as in step 516. If the present speech processing rate in step 516 has not caught up, the catch-up mechanism (s) are continued. If the speech processing rate in step 516 has caught up, the speech signal is transmitted and/or received at the normal rate, under normal conditions.

It will be understood that the communication unit and method for providing a communication link, as described above, provides at least the advantage that the opening

portion of speech, when performing a call set-up operation, is not truncated.

Furthermore, once a buffering operation has been performed on the speech signal to be transmitted, a number of alternative, or complementary, techniques have been provided to allow the processing of the buffered speech to catch up with the input speech signal, thereby minimizing any steady state delay. In utilising at least one of the aforementioned speech catch-up techniques, the length of time of active calls is reduced, thereby providing a more optimal use of the communication resource.

Although the invention has been described with reference to the DIIS communication specification, the inventive concepts contained herein are clearly suitable to alternative fixed or wireless communications system technologies and access protocols.

It is also within the contemplation of the invention that alternative arrangements for adapting the operation of speech processing units would benefit from the inventive concepts described herein. The invention is therefore not limited to use of a preferred dual-rate vocoder. A skilled person would readily recognise that the inventive concepts described herein could beneficially be used within a system that operated a variable speech coding or multi-rate coding process.

Thus a communication unit, a method of providing a communication link that avoids truncation of a speech signal and a communication system, adapted to facilitate the novel and inventive concepts of the communication unit and method, have been provided wherein the aforementioned truncated speech problems have been substantially alleviated.

Claims (16)

Claims

1. A communication unit comprises a speech processing unit operably coupled to a memory element and a processor, wherein the processor determines whether a communication link for the communication unit has been established, the communication unit characterised in that the speech processing unit receives a speech signal and determines whether to buffer, in the memory element, a first portion of said speech signal for subsequent transmittal, dependent upon said determination of whether a communication link has been established.

2. A communication unit according to claim 1, wherein the subsequent transmittal of the buffered first portion of the speech (404) is performed substantially upon establishing said communication link.

3. A communication unit according to claim 1 or claim 2, the communication unit having a speech input port for receiving said speech signal, the communication unit further characterised by said buffered first portion of the speech signal being processed by said speech processing unit at a faster rate than said speech signal input to said speech input port.

4. A communication unit according to claim 3, wherein said communication unit includes a multi-rate speech coder and the processing of buffered speech is performed using a rate of the multi-rate speech coder faster than a rate of said input speech signal.

5. A communication unit according to claim 4, wherein said multi-rate speech coder includes a linear predictive codebook and a fixed excitation codebook for each of the multi-speech processing rates provided by the multi-rate speech coder, a particular linear predictive codebook and a fixed excitation codebook being selected to process said buffered speech dependent upon said faster rate.

6. A communication unit according to any one of the preceding claims wherein the subsequent transmittal of the speech signal, after buffering of the initial portion of the input speech signal, is performed in a manner such that said subsequent transmitted speech signal aims to catch up with the input speech signal, the mechanism for catching up being performed by at least one of the following: compression of a pulse coded modulation aspect of the speech signal, silence compression of the speech signal, adapting a speech coding rate of the buffered speech, adapting a forward error correction encoding rate of the buffered speech.

7. A communication unit according to claim 6, the communication unit comprising first determination means to determine and adapt said speech coding rate of the buffered speech, and/or second determination means to determine and adapt said forward error correction encoding rate of the buffered speech.

8. A communication unit according to claim 7, the communication unit comprising third determination means to determine a data rate transmission rate and adapt said speech coding rate and/or forward error correction encoding rate by adapting a modulation rate of the communication unit.

9. A communication unit according to any one of the preceding claims, the communication unit further comprising a timer operably coupled to the processor and a controller for initiating processing of the buffered first portion of the speech signal once a communication link has been established.

10. A simplex communication system, semi-duplex communication system, radio communication system or fixed communication system having at least one processing element adapted to accommodate said buffering and subsequent transmission of speech signals from a communication unit in accordance with any one of the preceding claims.

11. A method for providing a communication link to a communication unit in a communication system, the method comprising the steps of: initiating a call set-up process by the communication unit; uttering speech, by the user, into a speech input port of the communication unit; determining whether a communication link has been established; buffering an initial portion of speech uttered by the user, upon determination that a communication link has not been established.

12. The method for providing a communication link to a communication unit in accordance with claim 11, the method further comprising the steps of: determining when a communication link has been established; and processing speech using one or more speech catch-up mechanism (s).

13. The method for providing a communication link to a communication unit in accordance with claim 12, wherein the catch-up mechanism (s) include at least one of the following steps: using a higher-rate speech coder than the true speech rate coder in a dual-or multi-rate coder, using time (PCM and/or silence) compression, adapting a FEC encoder rate; or adapting a speech coding rate.

14. The method for providing a communication link to a communication unit in accordance with any one of claims 11 to 13, the method further comprising the steps of: determining, by both a transmitting unit and a receiving unit, as to whether the speech processing rate has caught up with the input speech rate; and processing the speech at a normal rate, under normal conditions, if the speech processing rate has caught up.

15. A communication unit substantially as hereinbefore described with reference to, and/or as illustrated by, FIG. 1 or FIG. 2 or FIG. 3 of the accompanying drawings.

16. A method for providing a communication link to a communication unit substantially as hereinbefore described with reference to, and/or as illustrated by, FIG. 5 of the accompanying drawings.