Transmission of comfort noise parameters during discontinuous transmission

A comfort noise block, that include a hangover period and comfort noise parameters, is transmitted in such a manner that it is not interrupted by other messages, such as FACCH messages. This is accomplished in a mobile station by a determination of whether any FACCH messages are required to be transmitted. If such FACCH messages exist, a further determination may be made as to which transmission can be made in the shortest time (i.e., the FACCH message or messages or the comfort noise parameters message), and this transmission is made first. In any event the comfort noise parameters block is transmitted without interruption. In a further embodiment of this invention the comfort noise parameters message is transmitted by being concatenated with another message, such as a neighbor channel measurement results message, so as to reduce overhead, conserve bandwidth, and reduce power consumption. An element of the comfort noise parameters message is a Random Excitation Spectral Control (RESC) information element, which is used in the decoder for improving the spectral content of the generated comfort noise so as to better match the background noise at the transmitter.

FIELD OF THE INVENTION
 This invention relates generally to the field of speech communication, and
 more particularly to discontinuous transmission (DTX) and improving the
 quality of comfort noise (CN) during discontinuous transmission.
 BACKGROUND OF THE INVENTION
 Discontinuous transmission is used in mobile communication systems to
 switch the radio transmitter off during speech pauses. The use of DTX
 saves power in the mobile station and increases the time required between
 battery recharging. It also reduces the general interference level and
 thus improves transmission quality.
 However, during speech pauses the background noise which is transmitted
 with the speech also disappears if the channel is cut off completely. The
 result is an unnatural sounding audio signal (silence) at the receiving
 end of the communication.
 It is known in the art, instead of completely switching the transmission
 off during speech pauses, to instead generate parameters that characterize
 the background noise, and to send these parameters over the air interface
 at a low rate in Silence Descriptor (SID) frames. These parameters are
 used at the receive side to regenerate background noise which reflects, as
 well as possible, the spectral and temporal content of the background
 noise at the transmit side. These parameters that characterize the
 background noise are referred to as comfort noise (CN) parameters. The
 comfort noise parameters typically include a subset of speech coding
 parameters: in particular synthesis filter coefficients and gain
 parameters.
 It should be noted, however, that in some comfort noise evaluation schemes
 of some speech codecs, part of the comfort noise parameters are derived
 from speech coding parameters while other comfort noise parameter(s) are
 derived from, for example, signals that are available in the speech coder
 but that are not transmitted over the air interface.
 It is assumed in prior-art DTX systems that the excitation can be
 approximated sufficiently well by spectrally flat noise (i.e., white
 noise). In prior art DTX systems, the comfort noise is generated in the
 receiver by feeding locally generated, spectrally flat noise through a
 speech coder synthesis filter.
 Before describing the present invention, it will be instructive to review
 conventional circuitry and methods for generating comfort noise parameters
 on the transmit side, and for generating comfort noise on the receive
 side. In this regard reference is thus first made to FIGS. 1a-1d.
 Referring to FIG. 1a, short term spectral parameters 102 are calculated
 from a speech signal 100 in a Linear Predictive Coding (LPC) analysis
 block 101. LPC is a method well known in the prior art. For simplicity,
 discussed herein is only the case where the synthesis filter has only a
 short term synthesis filter, it being realized that in most prior art
 systems, such as in GSM FR, HR and EFR coders, the synthesis filter is
 constructed as a cascade of a short term synthesis filter and a long term
 synthesis filter. However, for the purposes of this description a
 discussion of the long term synthesis filter is not necessary.
 Furthermore, the long term synthesis filter is typically switched off
 during comfort noise generation in prior art DTX systems.
 The LPC analysis produces a set of short term spectral parameters 102 once
 for each transmission frame. The frame duration depends on the system. For
 example, in all GSM channels the frame size is set at 20 milliseconds.
 The speech signal is fed through an inverse filter 103 to produce a
 residual signal 104. The inverse filter is of the form:
 ##EQU1##
 The filter coefficients a(i), i=1, . . . , M are produced in the LPC
 analysis and are updated once for each frame. Interpolation as known in
 prior art speech coding may be applied in the inverse filter 103 to obtain
 a smooth change in the filter parameters between frames. The inverse
 filter 103 produces the residual 104 which is the optimal excitation
 signal, and which generates the exact speech signal 100 when fed through
 synthesis filter 1/A(z) 112 on the receive side (see FIG. 1b). The energy
 of the excitation sequence is measured and a scaling gain 106 is
 calculated for each transmission frame in excitation gain calculation
 block 105.
 The excitation gain 106 and short term spectral coefficients 102 are
 averaged over several transmission frames to obtain a characterization of
 the average spectral and temporal content of the background noise. The
 averaging is typically carried out over four frames for the GSM FR channel
 to eight frames, as is the case for the GSM EFR channel. The parameters to
 be averaged are buffered for the duration of the averaging period in
 blocks 107a and 108a (see FIG. 1d). The averaging process is carried out
 in blocks 107 and 108, and the average parameters that characterize the
 background noise are thus generated. These are the average excitation gain
 g.sub.mean and the average short term spectral coefficients. In modern
 speech codecs, there are typically 10 short term spectral coefficients
 (M=10) which are usually represented as Line Spectral Pair (LSP)
 coefficients f.sub.mean (i), i=1, . . . , M, as in the GSM EFR DTX system.
 Although these parameters are typically quantized prior to transmission,
 the quantization is ignored in this description for simplicity, in that
 the exact type of quantization that is performed is irrelevant to the
 teachings of this invention.
 Referring briefly to FIG. 1d, it is shown that the averaging blocks 107 and
 108 each typically include the respective buffers 107a and 108a, which
 output buffered signals 107b and 108b, respectively, to the averaging
 blocks.
 The computation and averaging of the comfort noise parameters is explained
 in detail in GSM recommendation: GSM 06.62 "Comfort noise aspects for
 Enhanced Full Rate (EFR) speech traffic channels". Also by example,
 discontinuous transmission is explained in GSM recommendation: GSM 06.81
 "Discontinuous Transmission (DTX) for Enhanced Full Rate (EFR) for speech
 traffic channels", and voice activity detection (VAD) is explained in GSM
 recommendation: GSM 06.82 "Voice Activity Detection (VAD) for Enhanced
 Full rate (EFR) speech channels". As such, the details of these various
 functions are not further discussed here.
 Referring to FIG. 1b, there is shown a block diagram of a conventional
 decoder on the receive side that is used to generate comfort noise in the
 prior art speech communication system. The decoder receives the two
 comfort noise parameters, the average excitation gain g.sub.mean and the
 set of average short term spectral coefficients f.sub.mean (i), i=1, . . .
 , M, and based on the parameters the decoder generates the comfort noise.
 The comfort noise generation operation on the receive side is similar to
 speech decoding, except that the parameters are used at a significantly
 lower rate (e.g., once every 480 milliseconds, as in the GSM FR and EFR
 channels), and no excitation signal is received from the speech encoder.
 During speech decoding the excitation on the receive side is obtained from
 a codebook that contains a plurality of possible excitation sequences, and
 an index for the particular excitation vector in the codebook is
 transmitted along with the other speech coding parameters. For a detailed
 description of speech decoding and the use of codebooks reference can be
 had to, by example, U.S. Pat. No.: 5,327,519, entitled "Pulse Pattern
 Excited Linear Prediction Voice Coder", by Jari Hagqvist, Kari Jarvinen,
 Kari-Pekka Estola, and Jukka Ranta, the disclosure of which is
 incorporated by reference herein in its entirety.
 During comfort noise generation, however, no index to the codebook is
 transmitted, and the excitation is obtained instead from a random number
 or excitation (RE) generator 110. The RE generator 110 generates
 excitation vectors 114 having a flat spectrum. The excitation vectors 114
 are then scaled by the average excitation gain g.sub.mean in scaling unit
 115 so that their energy corresponds to the average gain of the excitation
 104 on the transmit side. A resulting scaled random excitation sequence
 111 is then input to the speech synthesis filter 112 to generate the
 comfort noise 113. The average short term spectral coefficients f.sub.mean
 (i) are used in the speech synthesis filter 112.
 FIG. 1c illustrates the spectrum associated with the signal in different
 parts of the prior art decoder of FIG. 1b. The RE-generator 110 produces
 the random number excitation sequences 114 (and the scaled excitation 111)
 having a flat spectrum. This spectrum is shown by curve A. The speech
 synthesis filter 112 then modifies the excitation to produce a non-flat
 spectrum as shown in curve B.
 During a hangover period, or time between when a voice activity detector
 (VAD) indicates that speech has stopped and when the transmission is
 actually terminated, the speech coding parameters characterizing
 background noise are stored and averaged for constructing CN parameters.
 Reference in this regard can be had to FIGS. 3 and 4, which are exemplary
 of the GSM system. Since the VAD has detected speech inactivity, it is
 guaranteed that the speech frames contain only noise (and not speech), and
 thus these hangover frames can be used for the averaging of speech encoder
 parameters to evaluate the comfort noise parameters.
 The length of the hangover period is determined by the length of the SID
 averaging period, i.e., the length of the hangover period must be long
 enough to complete the averaging of the parameters before the resulting
 comfort noise parameters are to be transmitted in a SID frame. In the DTX
 system of the GSM full rate speech coder, the length of the hangover
 period equals four frames (the length of the SID averaging period), since
 the comfort noise evaluation technique uses only parameters from the
 previous frames to make an updated SID frame available. In the DTX system
 of the GSM enhanced full rate speech coder, the length of the hangover
 period equals seven frames (the length of the SID averaging period minus
 one), since the parameters of the eighth frame of the SID averaging period
 can be obtained from the speech encoder while processing the first SID
 frame. FIG. 3 illustrates the concepts of the hangover period and the SID
 averaging periods in the DTX system of the GSM enhanced full rate speech
 coder, and FIG. 4 shows as an example the longest possible speech burst
 without hangover.
 At the end of the hangover period the first SID frame is transmitted, and
 the comfort noise evaluation algorithm continues evaluating the
 characteristics of the background noise and passes the updated SID frames
 to the transmitter frame by frame, as long as the VAD continues to detect
 speech inactivity.
 It can be appreciated that, if the transmission of comfort noise parameters
 is not regular in nature, the resulting generated comfort noise may not
 match the original background noise at the transmitter.
 It can be further appreciated that if the comfort noise parameters are
 transmitted as separate, discrete messages, that a certain amount of
 system bandwidth is consumed. By example, if in the IS-136 system the CN
 parameters were sent in a dedicated Fast Associated Control Channel
 (FACCH) message, then two time slots would be required because of the two
 burst interleaving that is employed for FACCH messages.
 In the IS-136 system the FACCH is defined to be a blank and burst channel
 used for signalling exchange between the base station and the mobile
 station. A Slow Associated Control Channel (SACCH) is defined to be a
 continuous channel used for message exchange between the base station and
 the mobile station. A fixed number of bits are allocated to the SACCH in
 each TDMA slot.
 In the prior art GSM system the comfort noise parameters are sent in-band
 (i.e., coded into voice coder slots). While this technique may be
 applicable to other digital cellular standards, it would not be compatible
 with a presently specified IS-136 Enhanced Full Rate (EFR) voice coder. It
 has also been found that the approximately 0.5 second CN update that is
 performed in GSM may be relaxed, thereby utilizing less system bandwidth
 for CN updates.
 OBJECTS AND ADVANTAGES OF THE INVENTION
 It is thus a first object and advantage of this invention to provide an
 improved method for transmitting a comfort noise block during DTX
 operation.
 It is a further object and advantage of this invention to transmit a
 comfort noise block in such a manner that it is not interrupted by other
 messages, such as FACCH messages.
 It is one further object and advantage of this invention to concatenate a
 comfort noise parameter message with another message, such as a neighbor
 channel measurement results message, so as to reduce overhead, conserve
 bandwidth, and reduce power consumption.
 SUMMARY OF THE INVENTION
 The foregoing and other problems are overcome and the objects and
 advantages of the invention are realized by methods and apparatus in
 accordance with embodiments of this invention, wherein an improved method
 is provided for transmitting a comfort noise (CN) block, comprised of a
 hangover period and comfort noise parameters, during a discontinuous
 transmission (DTX) mode of operation.
 In accordance with the teaching of this invention the comfort noise block
 is transmitted in such a manner that it is not interrupted by other
 messages, such as FACCH messages. This is accomplished in the mobile
 station by a determination of whether any control channel messages, such
 as FACCH messages, are required to be transmitted. If such control channel
 messages exist, the mobile station groups or otherwise organizes the
 control channel message or messages such that a comfort noise block can be
 scheduled to be transmitted without interruption.
 In an embodiment of this invention, and if such FACCH messages exist, a
 further determination can be made as to which transmission can be made in
 the shortest time (i.e., the FACCH message or messages or the comfort
 noise block), and this transmission is made first.
 In a further embodiment of this invention the comfort noise parameters are
 transmitted by being concatenated with another message, such as a neighbor
 channel measurement results message, so as to reduce overhead, conserve
 bandwidth, and reduce power consumption.
 An element of the comfort noise parameters is a Random Excitation Spectral
 Control (RESC) information element, which is used in the decoder for
 improving the spectral content of the generated comfort noise so as to
 better match the background noise at the transmitter.

DETAILED DESCRIPTION OF THE INVENTION
 Reference is made to FIGS. 5 and 6 for illustrating a wireless user
 terminal or mobile station 10, such as but not limited to a cellular
 radiotelephone or a personal communicator, that is suitable for practicing
 this invention. The mobile station 10 includes an antenna 12 for
 transmitting signals to and for receiving signals from a base site or base
 station 30. The base station 30 is a part of a cellular network that may
 include a Base Station/Mobile Switching Center/Interworking function (BMI)
 32 that includes a mobile switching center (MSC) 34. The MSC 34 provides a
 connection to landline trunks when the mobile station 10 is involved in a
 call. In the context of this disclosure the mobile station 10 may be
 referred to as the transmission side and the base station as the receive
 side. The base station 30 is assumed to include suitable receivers and
 speech decoders for receiving and processing encoded speech parameters and
 also DTX comfort noise parameters, as described below.
 The mobile station includes a modulator (MOD) 14A, a transmitter 14, a
 receiver 16, a demodulator (DEMOD) 16A, and a controller 18 that provides
 signals to and receives signals from the transmitter 14 and receiver 16,
 respectively. These signals include signalling information in accordance
 with the air interface standard of the applicable cellular system, and
 also user speech and/or user generated data. The air interface standard is
 assumed for this invention to include a physical and logical frame
 structure, although the teaching of this invention is not intended to be
 limited to any specific structure, or for use only with an IS-136
 compatible mobile station, or for use only in TDMA type systems. The air
 interface standard is also assumed to support a DTX mode of operation.
 It is understood that the controller 18 also includes the circuitry
 required for implementing the audio and logic functions of the mobile
 station. By example, the controller 18 may be comprised of a digital
 signal processor device, a microprocessor device, and various analog to
 digital converters, digital to analog converters, and other support
 circuits. The control and signal processing functions of the mobile
 station are allocated between these devices according to their respective
 capabilities.
 A user interface includes a conventional earphone or speaker 17, a speech
 transducer such as a conventional microphone 19 in combination with an A/D
 converter and a speech encoder, a display 20, and a user input device,
 typically a keypad 22, all of which are coupled to the controller 18. The
 keypad 22 includes the conventional numeric (0-9) and related keys (#,*)
 22a, and other keys 22b used for operating the mobile station 10. These
 other keys 22b may include, by example, a SEND key, various menu scrolling
 and soft keys, and a PWR key. The mobile station 10 also includes a
 battery 26 for powering the various circuits that are required to operate
 the mobile station.
 The mobile station 10 also includes various memories, shown collectively as
 the memory 24, wherein are stored a plurality of constants and variables
 that are used by the controller 18 during the operation of the mobile
 station. For example, the memory 24 stores the values of various cellular
 system parameters and the number assignment module (NAM). An operating
 program for controlling the operation of controller 18 is also stored in
 the memory 24 (typically in a ROM device). The memory 24 may also store
 data, including user messages, that is received from the BMI 32 prior to
 the display of the messages to the user.
 It should be understood that the mobile station 10 can be a vehicle mounted
 or a handheld device. It should further be appreciated that the mobile
 station 10 can be capable of operating with one or more air interface
 standards, modulation types, and access types. By example, the mobile
 station may be capable of operating with any of a number of other
 standards besides IS-136, such as GSM. It should thus be clear that the
 teaching of this invention is not to be construed to be limited to any one
 particular type of mobile station or air interface standard. The operating
 program in the memory 24 includes routines to present messages and
 message-related functions to the user on the display 20, typically as
 various menu items. The memory 24 also includes routines for implementing
 the methods described below with regard to the transmission of comfort
 noise parameters during DTX operation.
 Although the invention is described next specifically in the context of an
 IS-136 embodiment, it is again noted that the teaching of this invention
 is not limited to only this one air interface standard.
 With regard to DTX on a digital traffic channel (IS-136.1, Rev. A, Section
 2.3.11.2), and as presently specified, when in the DTX-High state the
 transmitter 14 radiates at a power level indicated by the most recent
 power-controlling order (Initial Traffic Channel Designation message,
 Digital Traffic Channel (DTC) Designation message, Handoff message,
 Dedicated DTC Handoff message, or Physical Layer Control message) received
 by the mobile station 10.
 In the DTX-Low state, the transmitter 14 remains off. The CDVCC is not sent
 except for the transmission of FACCH messages. All Slow Associated Control
 Channel (SACCH) messages to be transmitted by the mobile station 10, while
 in the DTX-Low state, are sent as a FACCH message, after which the
 transmitter 14 returns again to the off state unless Discontinuous
 Transmission (DTX) has been otherwise inhibited.
 When the mobile station 10 desires to switch from the DTX-High state to the
 DTX-Low state, it may complete all in-progress SACCH messages in the
 DTX-High state, or terminate SACCH message transmission and resend the
 interrupted SACCH messages, in their entirety, as FACCH messages in the
 DTX-Low state.
 When a mobile station switches from the DTX High state to the DTX Low
 state, it must pass through a transition state in which the transmitted
 power is at the DTX High level until all pending FACCH messages have been
 entirely transmitted.
 In accordance with an aspect of this invention, the mobile station 10
 remains in the transition state until a Comfort Noise Block (comprised of
 six DTX hangover slots, and the related Comfort Noise Parameter message)
 have been entirely transmitted. The Comfort Noise Block is sent without
 interruption. If some other FACCH message slots coincide with the sending
 of the Comfort Noise Block, the mobile station 10 delays the transmission
 of either the FACCH message or the Comfort Noise Block so as to transmit
 one before the other, but in any case the FACCH messages are effectively
 grouped or segregated such that they do not interrupt or steal the slots
 used for the transmission of the Comfort Noise Block. This insures the
 best available quality of comfort noise that is generated at a base
 station voice/comfort noise decoder.
 In the mobile station 10, a determination is made by the controller 18 if
 there is a need to send hangover period slots, and if there is also a need
 to send any FACCH messages such as an acknowledgement type FACCH message
 of previously commanded channel quality measurement results (used for a
 mobile assisted handoff (MAHO) function). For example, the controller 18
 makes a determination as to the time required to send the comfort noise
 block and the time required to send the one or more FACCH messages. The
 transmission that can be achieved in the shortest amount of time is
 selected first, is transmitted, and then the other transmission (comfort
 noise block or FACCH message(s)) is made. Other criteria could also be
 employed, such as one based on message priority.
 In the case of a short speech/noise burst, only the Comfort Noise Parameter
 message is transmitted without the hangover slots. In this case there is
 no need to delay other coinciding FACCH messages.
 With regard to Mobile Assisted Handoff (MAHO) operations with DTX
 (IS-136.1, Rev. A, Sections 2.4.5.3 and 3.4.6.3), and as is presently
 specified, the mobile station 10 transmits the signal quality information
 over either the SACCH or the FACCH. In the case of continuous transmission
 (non-DTX), the mobile station 10 transmits over the SACCH. In the case of
 DTX, the mobile station 10 transmits channel quality information over the
 SACCH whenever the mobile station 10 is in the DTX high state. If the
 mobile station 10 is in the DTX low state, the data is sent from the
 mobile station 10 to the base station 30 by going to the DTX high state
 and transmitting the information over the FACCH.
 In accordance with a further aspect of this invention, when in the DTX low
 state, the CN Parameter message is appended or concatenated with the
 neighbor channel quality information sent over the FACCH. This technique
 thus avoids the use of separate FACCH messages to transmit the CN
 parameter message, and thus reduces overhead and conserves bandwidth and
 power.
 Furthermore, in the presently preferred embodiment of this invention the CN
 parameter message is sent at, by example, one second intervals from the
 mobile station 10 to the base station 30, thereby further reducing
 overhead. The one second interval in this case is related to the IS-136
 requirement that neighbor channel measurement results be reported to the
 base station 30 at one second intervals.
 It is also within the scope of the teaching of this invention to transmit
 the CN parameters, over the traffic channel, using DCCH channel coding and
 intra-slot interleaving. This can be used to enable the information to be
 sent in one slot. In this case the base station 30 determines if DCCH
 channel coding is being used, and reacts appropriately. This particular
 mode of operation is appropriate for when neighbor channel measurements
 are not in use.
 In accordance with a specific embodiment of this invention, the Comfort
 Noise (CN) Parameter Message, shown below in Table 1, is transmitted on
 the reverse digital traffic channel (RDTC), specifically the FACCH logical
 channel, and contains 38 bits, of which 26 bits contain a LSF residual
 vector which is quantized using the same split vector quantization (SVQ)
 codebook as used in the IS-641 speech codec. The
 quantization/dequantization algorithms of the speech codec are modified to
 make it possible to use this codebook. The LSF parameters give an estimate
 of the spectral envelope of the background noise at the transmit side
 using a 10th order LPC model of the spectrum.
 The next 8 bits contain a comfort noise energy quantization index, which
 describes the energy of the background noise at the transmit side. The
 remaining 4 bits in the message are used for transmitting a Random
 Excitation Spectral Control (RESC) information element.
 TABLE 1
 Message Format
 Information Element Type Length (bits)
 Protocol Discriminator M 2
 Message Type M 8
 LSF residual vector M 26
 CN energy quantization M 8
 index
 RESC parameters M 4
 The nature of the RESC information element can be better understood with
 reference to FIGS. 2a-2c. The conventional technique for both encoding and
 decoding comfort noise was described above. In FIGS. 2a and 2b those
 elements that appear also in FIGS. 1a and 1b are numbered accordingly.
 Referring now to FIG. 2a, there is shown a block diagram of apparatus for
 generating comfort noise parameters on transmit side. The RESC-related
 operations are separated from those known from the prior art by a dashed
 line 204. According to this technique, the residual signal 104 output from
 the inverse filter 103 is subjected to a further analysis (such as
 LPC-analysis) to produce another set of filter coefficients. The second
 analysis, which is referred to herein as random excitation (RE)
 LPC-analysis 200, is typically of a lower degree than the LPC analysis
 carried out in block 101. The RE LPC-analysis block 200 produces random
 excitation spectral control parameters r.sub.mean (i), i=1, . . . ,R. The
 parameters are obtained by averaging the spectral parameters 201 from the
 RE LPC-analysis block 200 over several consecutive frames in averaging
 block 203. The RESC parameters characterize the spectrum of the
 excitation.
 It should be noted that the RESC parameters are not a subset of the speech
 coding parameters, but are generated and used only during comfort noise
 generation. The inventors have found that first or second order
 LPC-analysis is sufficient to generate the RESC parameters (R=1 or 2).
 However, spectral models other than the all-pole model of the LPC
 technique may also be used. The averaging may alternatively be carried out
 by the RE LPC analysis block 200 by averaging the autocorrelation
 coefficients within the LPC parameter calculation, or by any other
 suitable averaging means within the LPC coefficient computation. The
 averaging period for the RESC parameters may be the same as that used for
 the other CN parameters, but is not restricted to only the same averaging
 period. For example, it has been found that longer averaging, than what is
 used for the conventional CN-parameters, can be advantageous. Thus,
 instead of using an averaging period of seven frames, a longer averaging
 period may be preferred (e.g., 10-12 frames).
 Prior to calculating the excitation gain, the LPC-residual 104 is fed
 through a second inverse filter H.sub.RESC (Z) 202. This filter produces a
 spectral controlled residual 205 which generally has a flatter spectrum
 than the LPC-residual 104. The random excitation spectral control (RESC)
 inverse filter H.sub.RESC (z) may be of the form of an all-zero filter
 (but not restricted to only this form):
 ##EQU2##
 The excitation gain is calculated from the spectrally flattened residual
 205. Otherwise the operations in FIG. 2a are similar to those described
 above with regard to FIG. 1a.
 The RESC parameters, along with the other CN parameters, are then
 transmitted from the mobile station 10 using the techniques described
 above with regard to the FACCH and the MAHO related operations when DTX is
 active.
 Referring now to FIG. 2b, there is shown a block diagram of decoder on the
 receive side that is used to generate comfort noise according to the
 present invention. In the decoder, the excitation 212 is formed by first
 generating the white noise excitation sequence 114 with the random
 excitation generator 110, which is then scaled by g.sub.mean in scaling
 block 115.
 The spectrally flat noise sequence 111 is then processed in a random
 excitation spectral control (RESC) filter 211, which produces an
 excitation having a correct spectral content. The RE spectral control
 filter 211 performs the inverse operation to the RESC inverse filter 202
 employed in the encoder of FIG. 2a. Using the RESC inverse filter of
 equation (2) on the transmit side, the RE spectral control filter 211 used
 on the receive side is of the form
 ##EQU3##
 The RESC-parameters r.sub.mean (i), i=1, . . . , R that define the filter
 coefficients b(i), i=1, . . . , R are transmitted as part of the CN
 parameters to the receive side, and are used in the RE spectral control
 filter 211 so that the excitation for the synthesis filter 112 is suitably
 spectrally weighted, and is thus generally not flat spectrum. The RESC
 parameters r.sub.mean (i), i=1, . . . , R may be the same as the filter
 coefficients b(i), i=1, . . . , R, or they may use some other parameter
 representation that enables efficient quantization for transmission, such
 as LSP coefficients. FIGS. 7a-7g illustrate exemplary frequency responses
 of the RESC filter 211.
 In review, the CN-excitation generator 210 generates a spectrally flat
 random excitation in the RE generator 110. The spectrally flat excitation
 is then suitably scaled by the average gain scaler 115. To produce the
 correct spectrum, and to avoid a mismatch between the spectrum of the
 comfort noise and that of the background noise, the random excitation is
 fed through the RE spectrum control filter 211. The spectrally controlled
 excitation 212 is then used in the speech synthesis filter 112 to produce
 comfort noise that has an improved match to the spectrum of the actual
 background noise that is present at the transmit side.
 The RESC parameters are not a subset of the speech coding parameters that
 are used during speech signal processing, but are instead calculated only
 during the comfort noise calculation. The RESC parameters are computed and
 transmitted only for the purpose of generating improved excitation for
 comfort noise during speech pauses. The RESC inverse filter 202 in the
 encoder and the RESC filter 211 in the decoder are used only for the
 purpose of controlling the spectrum of the random excitation.
 FIG. 2c illustrates the spectrum of certain signals within the decoder of
 FIG. 2b during the generation of comfort noise according to the present
 invention. The RE generator 110 produces the random number sequences
 having the flat spectrum shown in curve A. This spectrum is identical to
 the curve A shown in 120 of FIG. 1c. Signals 114 and 111 both have this
 flat spectrum, it being noted that the gain scaling that occurs in block
 115 does not affect the shape of the spectrum. The white noise sequence
 111 is then fed through RE spectrum control filter 211 to produce the
 excitation 212 to the LPC synthesis filter. The improved excitation
 sequence 212 generally has a non-flat spectrum (curve C), and the effect
 of this non-flat spectrum is observed in the output spectrum (curve D) of
 the synthesis filter 112. The excitation sequence 212 may be lowpass or
 highpass type, or may exhibit a more sophisticated frequency content
 (depending on the degree of the RESC filter). The spectrum control is
 determined by the RESC parameters, which are computed on the transmit side
 and transmitted as part of comfort noise to the receive side, as was
 described above.
 As was stated above, the Discontinuous Transmission (DTX) is a mechanism
 which allows the radio transmitter to be switched off most of the time
 during speech pauses for at least the purposes of saving power in the
 mobile station 10 and reducing the overall interference level in the air
 interface. DTX may be active in an IS-136 compatible mobile station 10 if
 allowed by the network, see IS-136.2, Section 2.6.5.2.
 The problems discussed in the Background section of this patent application
 are addressed by generating, on the receive side, a synthetic noise
 similar to the transmit side background noise. The comfort noise (CN)
 parameters ar estimated on the transmit side and transmitted to the
 receive side before the radio transmission is switched off, and at a
 regular low rate afterwards. This allows the comfort noise to adapt to the
 changes of the noise on the transmit side. The DTX mechanism in accordance
 with this invention employs: the Voice Activity Detector (VAD) 21 (FIG. 5)
 on the transmit side; an evaluation of the background acoustic noise on
 the transmit side, in order to transmit characteristic parameters to the
 receive side; and a generation on the receive side of a similar noise,
 referred to as comfort noise, during periods where the radio transmission
 is switched off.
 In addition to these functions, if the parameters arriving at the receive
 side are found to be seriously corrupted by errors, the speech or comfort
 noise is instead generated from substituted data in order to avoid
 generating annoying audio effects for the listener.
 The transmit side DTX function continuously passes traffic frames, each
 marked by a flag SP, to the radio transmitter 14, where the SP flag="1"
 indicates a speech frame, and where the SP flag="0" indicates an encoded
 set of Comfort Noise parameters. The scheduling of the frames for
 transmission on the air interface is controlled by the radio transmitter
 14, on the basis of the SP flag.
 In a preferred embodiment of this invention, and to allow an exact
 verification of the transmit side DTX functions, all frames before the
 reset of the mobile station 10 are treated as if they were speech frames
 for an infinitely long time. Therefore, the first 6 frames after the reset
 are always marked with SP flag="1", even if VAD flag="0" (hangover period,
 see FIG. 8).
 The Voice Activity Detector (VAD) 21 operates continuously in order to
 determine whether the input signal from the microphone 19 contains speech.
 The output is a binary flag (VAD flag="1" or VAD flag="0", respectively)
 on a frame by frame basis.
 The VAD flag controls indirectly, via the transmit side DTX handler
 operations described below, the overall DTX operation on the transmit
 side.
 Whenever the VAD flag="1", the speech encoded output frame is passed
 directly to the radio transmitter 14, marked with the SP flag="1".
 At the end of a speech burst (transition VAD flag="1" to VAD flag="0"), it
 requires seven consecutive frames to make a new updated set of CN
 parameters available. Normally, the first six speech encoder output frames
 after the end of the speech burst are passed directly to the radio
 transmitter 14, marked with the SP flag="1", thereby forming the "hangover
 period". The first new set of CN parameters is then passed to the radio
 transmitter 14 as the seventh frame after the end of the speech burst,
 marked with the SP flag="0" (see FIG. 8).
 If, however, at the end of the speech burst, less than 24 frames have
 elapsed since the last set of CN parameters were computed and passed to
 the radio transmitter 14, then the last set of CN parameters are
 repeatedly passed to the radio transmitter 14, until a new updated set of
 CN parameters is available (seven consecutive frames marked with VAD
 flag="0"). This reduces the activity on the air interface in cases where
 short background noise spikes are interpreted as speech, by avoiding the
 "hangover" waiting for the CN parameter computation. FIG. 9 shows as an
 example the longest possible speech burst without hangover.
 Once the first set of CN parameters after the end of a speech burst has
 been computed and passed to the radio transmitter 14, the transmit side
 DTX handler continuously computes and passes updated sets of CN parameters
 to the radio transmitter 14, marked with the SP flag="0", so long as the
 VAD flag="0".
 The speech encoder is operated in a normal speech encoding mode if the SP
 flag="1" and in a simplified mode if the SP flag="0", because not all
 encoder functions are required for the evaluation of CN parameters.
 In the radio transmitter 14 the following traffic frames are scheduled for
 transmission: all frames marked with the SP flag="1"; the first frame
 marked with the SP flag="0" after one or more frames with the SP flag="1";
 those frames marked with SP="0" and aligned with the transmission
 instances of the channel quality information sent over the FACCH.
 This has the overall effect that the radio transmission is terminated after
 the transmission of a FACCH CN parameter message when the speaker stops
 talking. During speech pauses the transmission is resumed at regular
 intervals for transmission of one FACCH CN parameter message, in order to
 update the generated comfort noise on the receive side (and to provide
 updated measurement results of the channel quality).
 The comfort noise evaluation algorithm uses the unquantized and quantized
 Linear Prediction (LP) parameters of the speech encoder, using the Line
 Spectral Pair (LSP) representation, where the unquantized Line Spectral
 Frequency (LSF) vector is given by f.sup.t =[f.sub.1 f.sub.2. . .f.sub.10
 ] and the quantized LSF vector by f.sup.t =[f.sub.1 f.sub.2. . . f.sub.10
 ], with t denoting transpose. The algorithm also uses the LP residual
 signal r(n) of each subframe for computing the random excitation gain and
 the Random Excitation Spectral Control (RESC) parameters.
 The algorithm computes the following parameters to assist in comfort noise
 generation: the reference LSF parameter vector f.sup.ref (average of the
 quantized LSF parameters of the hangover period); the averaged LSF
 parameter vector f.sup.mean (average of the LSF parameters of the seven
 most recent frames); the averaged random excitation gain g.sub.cn.sup.mean
 (average of the random excitation gain values of the seven most recent
 frames); the random excitation gain g.sub.cn ; and the RESC parameters
 .LAMBDA..
 These parameters give information on the spectrum
 (f,f,f.sup.ref,f.sup.mean,.LAMBDA.) and the level (g.sub.cn,
 g.sub.cn.sup.mean) of the background noise.
 Three of the evaluated comfort noise parameters (f.sup.mean,.LAMBDA., and
 g.sub.cn.sup.mean) are encoded into a special FACCH message, referred to
 herein as the Comfort Noise (CN) parameter message, for transmission to
 the receive side. Since the reference LSF parameter vector f.sup.ref can
 be evaluated in the same way in the encoder and decoder, as described
 below, no transmission of this parameter vector is necessary.
 The CN parameter message also serves to initiate the comfort noise
 generation on the receive side, as a CN parameter message is always sent
 at the end of a speech burst, i.e., before the radio transmission is
 terminated.
 The scheduling of CN parameter messages or speech frames on the radio path
 was described above with reference to FIGS. 8 and 9.
 The background noise evaluation involves computing three different kinds of
 averaged parameters: the LSF parameters, the random excitation gain
 parameter, and the RESC parameters. The comfort noise parameter to be
 encoded into a Comfort Noise parameter message are calculated over the CN
 averaging period of N=7 consecutive frames marked with VAD="0", as
 described in greater detail below.
 Prior to averaging the LSF parameters over the CN averaging period, a
 median replacement is performed on the set of LSF parameters to be
 averaged, to remove the parameters which are not characteristic of the
 background noise on the transmit side. First, the spectral distances from
 each of the LSF parameter vectors f(i) to the other LSF parameter vectors
 f(j), i=0, . . . 6, j=0, . . . , 6, i.noteq.j, within the CN averaging
 period are approximated according to the equation:
 ##EQU4##
 where f.sub.i (k) is the kth LSF parameter of the LSF parameter vector f(i)
 at frame i.
 To find the spectral distance .DELTA.S.sub.i of the LSF parameter vector
 f(i) to the LSF parameter vectors f(j) of all other frames j=0, . . . 6,
 j.noteq.i, within the CN averaging period, the sum of the spectral
 distances .DELTA.R.sub.ij is computed as follows:
 ##EQU5##
 for all i=0 . . . 6, i not equal to j.
 The LSF parameter vector f(i) with the smallest spectral distance
 .DELTA.S.sub.i of all the LSF parameter vectors within the CN averaging
 period is considered as the median LSF parameter vector f.sub.med of the
 averaging period, and its spectral distance is denoted as
 .DELTA.S.sub.med. The median LSF parameter vector is considered to contain
 the best representation of the short-term spectral detail of the
 background noise of all the LSF parameter vectors within the averaging
 period. If there are LSF parameter vectors f(j) within the CN averaging
 period with:
 ##EQU6##
 where TH.sub.med =2.25 is the median replacement threshold, then at most
 two of these LSF parameter vectors (the LSF parameter vectors causing
 TH.sub.med to be exceeded the most) are replaced by the median LSF
 parameter vector prior to computing the averaged LSF parameter vector
 f.sup.mean.
 The set of LSF parameter vectors obtained as a result of the median
 replacement are denoted as f'(n-i), where n is the index of the current
 frame, and i is the averaging period index (i=0 . . . 6).
 When the median replacement is performed at the end of the hangover period
 (first CN update), all of the LSF parameter vectors f(n-i) of the six
 previous frames (the hangover period, i=1 . . . 6) have quantized values,
 while the LSF parameter vector f(n) at the most recent frame n has
 unquantized values. In the subsequent CN update, the LSF parameter vectors
 of the CN averaging period in those frames overlapping with the hangover
 period have quantized values, while the parameter vectors of the more
 recent frames of the CN averaging period have unquantized values. If the
 period of the seven most recent frames is non-overlapping with the
 hangover period, the median replacement of LSF parameters is performed
 using only unquantized parameter values.
 The averaged LSF parameter vector f.sup.mean (n) at frame n is computed
 according to the equation:
 ##EQU7##
 where f'(n-i) is the LSF parameter vector of one of the seven most recent
 frames (i=0 . . . 6) after performing the median replacement, i is the
 averaging period index, and n is the frame index.
 The averaged LSF parameter vector f.sup.mean (n) at frame n is preferably
 quantized using the same quantization tables that are also used by the
 speech coder for the quantization of the non-averaged LSF parameter
 vectors in the normal speech encoding mode, but the quantization algorithm
 is modified in order to support the quantization of comfort noise. The LSF
 prediction residual to be quantized is obtained according to the following
 equation:
EQU r (n)=f.sup.mean (n)-f.sup.ref (8)
 where f.sup.mean (n) is the averaged LSF parameter vector at frame n,
 f.sup.ref is the reference LSF parameter vector, r(n) is the computed LSF
 prediction residual vector at frame n, and n is the frame index.
 The computation of the reference LSF parameter vector f.sup.ref is made on
 the basis of the quantized LSF parameters f by averaging these parameters
 over the hangover period of six frames according to the following
 equation:
 ##EQU8##
 where f(n-i) is the quantized LSF parameter vector of one of the frames of
 the hangover period (i=1 . . . 6), i is the hangover period frame index,
 and n is the frame index. It should be noted that the quantized LSF
 parameter vectors f(n-i) used for computing f.sup.ref are not subjected to
 median replacement prior to averaging.
 For each CN generation period the computation of the reference LSF
 parameter vector f.sup.ref is done only once at the end of the hangover
 period, and for the rest of the CN generation period f.sup.ref is frozen.
 The reference LSF parameter vector f.sup.ref is evaluated in the decoder
 in the same way as in the encoder, because during the hangover period the
 same LSF parameter vectors f are available at the encoder and decoder. An
 exception to this are the cases when transmission errors are severe enough
 to cause the parameters to become unusable, and a frame substitution
 procedure is activated. In these cases, the modified parameters obtained
 from the frame substitution procedure are used instead of the received
 parameters.
 The random excitation gain is computed for each subframe, based on the
 energy of the LP residual signal of the subframe, according to the
 following equation:
 ##EQU9##
 where g.sub.cn (j) is the computed random excitation gain of subframe j,
 r(l) is the lth sample of the LP residual of subframe j, and 1 is the
 sample index (l=0 . . . 39). The scaling factor of 1.286 is used to make
 the level of the comfort noise match that of the background noise coded by
 the speech codec. The use of this particular scaling factor value should
 not be read as a limitation of the practice of this invention.
 The computed energy of the LP residual signal is divided by the value of 10
 to yield the energy for one random excitation pulse, since during comfort
 noise generation the subframe excitation signal (pseudo noise) has 10
 non-zero samples, whose amplitudes can take values of +1 or -1.
 The computed random excitation gain values are averaged and updated in the
 first subframe of each frame n marked with VAD="0" according to the
 equation:
 ##EQU10##
 where g.sub.cn (n) (l) is the computed random excitation gain at the first
 subframe of frame n, g.sub.cn (n-i) (j) is the computed random excitation
 gain at subframe j of one of the past frames (i=1 . . . 6), and n is the
 frame index. Since the random excitation gain of only the first subframe
 of the current frame is used in the averaging, it is possible to make the
 updated set of CN parameters available for transmission after the first
 subframe of the current frame has been processed.
 The averaged random excitation gain is bounded by
 g.sub.cn.sup.mean.ltoreq.8064 and quantized with an 8-bit non-uniform
 algorithmic quantizer in the logarithmic domain, requiring no storage of a
 quantization table.
 With regard to the computation of RESC parameters, since the LP residual
 r(n) deviates somewhat from flat spectral characteristics, some loss in
 comfort noise quality (spectral mismatch between the background noise and
 the comfort noise) will result when a spectrally flat random excitation is
 used for synthesizing comfort noise on the receive side. To provide an
 improved spectral match, a further second order LP analysis is performed
 for the LP residual signal over the CN averaging period, and the resulting
 averaged LP coefficients are transmitted to the receive side in the CN
 parameter message to be used in the comfort noise generation. This method
 is referred to as the random excitation spectral control (RESC), and the
 obtained LP coefficients are referred to as the RESC parameters .LAMBDA..
 The LP residual signals r(n) of each subframe in a frame are concatenated
 to compute the autocorrelations r.sub.res (k), k=0 . . . 2, of the LP
 residual signal of the 20 ms frame according to the equation:
 ##EQU11##
 After computing the autocorrelations according to the foregoing equation,
 the autocorrelations are normalized to obtain the normalized
 autocorrelations r'.sub.res (k).
 For the most recent frame of the CN averaging period, the autocorrelations
 from only the first subframe are used for averaging to make it possible to
 prepare the updated set of CN parameters for transmission after the first
 subframe of the current frame has been processed.
 The computed normalized autocorrelations are averaged and updated in the
 first subframe of each frame n marked with VAD="0" according to the
 equation:
 ##EQU12##
 where r'.sub.res (n) (l) are the normalized autocorrelations at the first
 subframe of frame n, r'.sub.res (n-i) are the normalized autocorrelations
 of one of the past frames (i=1 . . . 6), and n is the frame index.
 The computed averaged autocorrelations r.sub.ref.sup.mean are input to a
 Schur recursion algorithm to compute the two first reflection
 coefficients, i.e., the RESC parameters .DELTA., or .lambda.(i), i=1, 2.
 Each of the two RESC parameters are encoded using a 2-bit scalar
 quantizer.
 The modification of the speech encoding algorithm during DTX operation is
 as follows. When the SP flag is equal to "0" the speech encoding algorithm
 is modified in the following way. The non-averaged LP parameters which are
 used to derive the filter coefficients of the short-term synthesis filter
 H(z) of the speech encoder are not quantized, and the memory of weighing
 filter W(z) is not updated, but rather set to zero. The open loop pitch
 lag search is performed, but the closed loop pitch lag search is
 inactivated and the adaptive codebook gain is set to zero. If the VAD
 implementation does not use the delay parameter of the adaptive codebook
 for making the VAD decision, the open loop pitch lag search can also be
 switched off. No fixed codebook search is performed. In each subframe the
 fixed codebook excitation vector of the normal speech decoder is replaced
 by a random excitation vector which contains 10 non-zero pulses. The
 random excitation generation algorithm is defined below. The random
 excitation is filtered by the RESC synthesis filter, as described below,
 to keep the contents of the past excitation buffer as nearly equal as
 possible in both the encoder and the decoder, to enable a fast startup of
 the adaptive codebook search when the speech activity begins after the
 comfort noise generation period. The LP parameter quantization algorithm
 of the speech encoding mode is inactivated. At the end of the hangover
 period the reference LSF parameter vector f.sup.ref is calculated as
 defined above. For the remainder of the comfort noise insertion period
 f.sup.ref is frozen. The averaged LSF parameter vector f.sup.mean is
 calculated each time a new set of CN parameters is to be prepared. This
 parameter vector is encoded into the CN parameter message was as defined
 above. The excitation gain quantization algorithm of the speech encoding
 mode is also inactivated. The averaged random excitation gain value
 g.sub.cn.sup.mean is calculated each time a new set of CN parameters is to
 be prepared. This gain value is encoded into the CN parameter message as
 previously defined. The computation of the random excitation gain is
 performed based on the energy of the LP residual signal, as defined above.
 The predictor memories of the ordinary LP parameter quantization and fixed
 codebook gain quantization algorithms are reset when the SP flag="0", so
 that the quantizers start from their initial states when the speech
 activity begins again. And finally, the computation of the RESC parameters
 is based on the spectral content of the LP residual signal, as defined
 above. The RESC parameters are computed each time a new set of CN
 parameters is to be prepared.
 The comfort noise encoding algorithm produces 38 bits for each CN parameter
 message as shown in Table 2. These bits are referred to as vector cn[0 . .
 . 37]. The comfort noise bits cn[0 . . . 37] are delivered to the FACCH
 channel encoder in the order presented in Table 2 (i.e., no ordering
 according to the subjective importance of the bits is performed).
 TABLE 2
 Detailed bit allocation of comfort noise parameters
 Index (vector to
 FACCH channel
 encoder) Description Parameter
 cn0-cn7 Index of 1st LSF VQ index of
 subvector r[1 . . . 3]
 cn8-cn16 Index of 2nd LSF VQ index of
 subvector r[4 . . . 6]
 cn17-cn25 Index of 3rd LSF VQ index of
 subvector r[7 . . . 10]
 cn26-cn33 Random excitation Index of g.sup.mean.sub.cn
 gain
 cn34-cn35 Index of 1st RESC Index of .lambda.(1)
 parameter
 cn36-cn37 Index of 2nd RESC Index of .lambda.(2)
 parameter
 Regardless of their context (speech, CN parameter message, other FACCH
 messages or none), the radio receiver of the base station 30 continuously
 passes the received traffic frames to the receive side DTX handler,
 individually marked by various preprocessing functions with three flags.
 These are the speech frame Bad Frame Indicator (BFI) flag, the comfort
 noise parameter Bad Frame Indicator (BFI CN) flag, and the Comfort Noise
 Update Flag (CNU) described below and in Table 3. These flags serve to
 classify the traffic frames according to their purpose. This
 classification, summarized in Table 3, allows the receive side DTX handler
 to determine in a simple way how the received frame is to be processed.
 TABLE 3
 Classification of traffic frames
 BFI CN
 BFI 0 1
 0 Unusable frame Good speech frame
 1 Valid CN parameter Unusable frame
 message
 The binary BFI and BFI CN flags indicate whether the traffic frame is
 considered to contain meaningful information bits (BFI flag="0" and BFI CN
 flag="1", or BFI flag="1" and BFI CN flag="0") or not (BFI flag="1" and
 BFI CN flag="1", or BFI flag="0" and BFI CN flag="0"). In the context of
 this disclosure, a FACCH frame is considered not to contain meaningful
 bits unless it contains a CN parameter message, and is thus marked with
 BFI flag="1" and BFI CN flag="1".
 The binary CNU flag marks with CNU="1" those traffic frames that are
 aligned with the transmission instances of the channel quality information
 sent over the FACCH.
 The receive side DTX handler is responsible for the overall DTX operation
 on the receive side. The DTX operation on the receive side is as follows:
 whenever a good speech frame is detected, the DTX handler passes it
 directly on to the speech decoder; when lost speech frames or lost CN
 parameter messages are detected, the substitution and muting procedure is
 applied; valid CN parameter messages frames result in comfort noise
 generation until the next CN parameter message is detected (CNU="1") or
 good speech frames are detected. During this period, the receive side DTX
 handler ignores any unusable frames delivered by the radio receiver; the
 parameters of the first lost CN parameter message are substituted by the
 parameters of the last valid CN parameter message and the procedure for
 the CN parameter message is applied; and upon reception of a second lost
 CN parameter message, muting is applied.
 With regard to the averaging and decoding of the LP parameters, when speech
 frames are received by the decoder the LP parameters of the last six
 speech frames are kept in memory. The decoder counts the number of frames
 elapsed since the last set of CN parameters was updated and passed to the
 radio transmitter by the encoder. Based on this count the decoder
 determines whether or not there is a hangover period at the end of the
 speech burst (if at least 30 frames have elapsed since the last CN
 parameter update when the first CN parameter message after a speech burst
 arrives, the hangover period is determined to have existed at the end of
 the speech burst).
 As soon as a CN parameter message is received, and the hangover period is
 detected at the end of the speech burst, the stored LP parameters are
 averaged to obtain the reference LSF parameter vector f.sup.ref. The
 reference LSF parameter vector and the reference fixed codebook gain value
 are frozen and used for the actual comfort noise generation period.
 The averaging procedure for obtaining the reference is as follows:
 When a speech frame is received, the LSF parameters are decoded and stored
 in memory. When the first CN parameter message is received, and the
 hangover period is detected at the end of the speech burst, the stored LSF
 parameters are averaged in the same way as in the speech encoder as
 follows:
 ##EQU13##
 where f(n-i) is the quantized LSF parameter vector of one of the frames of
 the hangover period (i=1 . . . 6), and n is the frame index.
 Once the reference LSF parameter vector has been computed, the averaged LSF
 parameter vector f.sup.mean (n) at frame n (encoded into the CN parameter
 message) can be reproduced at the decoder each time a CN update message is
 received according to the equation:
EQU f.sup.mean (n)=r(n)+f.sup.ref (15)
 where f.sup.mean (n) is the quantized averaged LSF parameter vector at
 frame n, f.sup.ref is the reference LSF parameter vector, r(n) is the
 received quantized LSF prediction residual vector at frame n, and n is the
 frame index.
 In each subframe, the fixed codebook excitation vector of the normal speech
 decoder containing four non-zero pulses is replaced during speech
 inactivity by a random excitation vector which contains 10 non-zero
 pulses. The pulse positions and signs of the random excitation are locally
 generated using uniformly distributed pseudo-random numbers. The
 excitation pulses take values of +1 and -1 in the random excitation
 vector. The random excitation generation algorithm operates in accordance
 with the following pseudo-code.

Pseudo-Code:
 for (i = 0; i &lt; 40; i++) code(i) = 0;
 for (i = 0; i &lt; 10; i++) {
 j = random (4);
 idx = j * 10 + i;
 if (random(2) == 1) code(idx) = 1;
 else code(idx) = -1;
 }
 where code [0 . . . 39] is the fixed codebook excitation buffer, and random
 (k) generates pseudo-random integer values, uniformly distributed over the
 range [0 . . . k-1].
 The received RESC parameter indices are decoded to obtain the received RESC
 parameters .lambda.(i), i=1,2. After the random excitation has been
 generated, it is filtered by the RESC synthesis filter, defined as
 follows:
 ##EQU14##
 The RESC synthesis filter is preferably implemented using a lattice
 filtering method. After RESC synthesis filtering, the random excitation is
 subjected to scaling and LP synthesis filtering.
 The comfort noise generation procedure uses the speech decoder algorithm
 with the following modifications. The fixed codebook gain values are
 replaced by the random excitation gain value received in the CN parameter
 message, and the fixed codebook excitation is replaced by the locally
 generated random excitation as was described above. The random excitation
 is filtered by the RESC synthesis filter, as was also described above. The
 adaptive codebook gain value in each subframe is set to 0. The pitch delay
 value in each subframe is set to, for example, 60. The LP filter
 parameters used are those received in the CN parameter message. The
 predictor memories of the ordinary LP parameter and fixed codebook gain
 quantization algorithms are reset when the SP flag="0", so that the
 quantizers start from their initial states when the speech activity begins
 again. With these parameters, the speech decoder now performs its standard
 operations and synthesizes comfort noise. Updating of the comfort noise
 parameters (random excitation gain, RESC parameters, and LP filter
 parameters) occurs each time a valid CN parameter message is received, as
 described above. When updating the comfort noise, the foregoing parameters
 are interpolated over the CN update period to obtain smooth transitions.
 A lost CN parameter message is defined as an unusable frame that is
 received when the receive side DTX handler is generating comfort noise and
 a CN parameter message is expected (Comfort Noise Update flag, CNU="1").
 The parameters of a single lost CN parameter message are substituted by the
 parameters of the last valid CN parameter message and the procedure for
 valid CN parameters is applied. For the second lost CN parameter message,
 a muting technique is used for the comfort noise that gradually decreases
 the output level (-3 dB/frame), resulting in eventual silencing of the
 output of the decoder. The muting is accomplished by decreasing the random
 excitation gain with a constant value of -3 dB in each frame down to a
 minimum value of 0. This value is maintained if additional lost CN
 parameter messages occur.
 Although a number of presently preferred embodiments of this invention have
 been described with respect to specific values of frame durations, numbers
 of frames, and the like, it should be realized that the numbers of frames,
 duration of frames, duration of the hangover period, duration of the
 averaging period, etc., may be varied in accordance with the
 specifications and requirements of different types of digital mobile
 communications systems. Furthermore, and although the invention has been
 described in the context of circuit block diagrams, it will be appreciated
 that some of the illustrated circuit blocks are implemented by a suitably
 programmed digital data processor that forms a portion of the digital
 cellular telephone.
 Thus, while the invention has been particularly shown and described with
 respect to preferred embodiments thereof, it will be understood by those
 skilled in the art that changes in form and details may be made therein
 without departing from the scope and spirit of the invention.