WO2001035394A2

WO2001035394A2 - Integrated voice and data transmission based on bit importance ranking

Info

Publication number: WO2001035394A2
Application number: PCT/US2000/041953
Authority: WO
Inventors: David Rand Irvin; Ali S. Khayrallah
Original assignee: Ericsson Inc.
Priority date: 1999-11-08
Filing date: 2000-11-07
Publication date: 2001-05-17
Also published as: AU3078501A; WO2001035394A3

Abstract

A system for integrating a digitized analog signal with a separate data signal to provide an integrated signal, transmitting the integrated signal, receiving the integrated signal and regenerating the analog signal and data signal. In one embodiment an analog voice signal is converted to a digital signal comprising a plurality of bands. The energy of each band is approximated and compared by means of an importance gauge to a predetermined threshold level in order to identify high energy bands. The information in the high energy bands is integrated with a separate data signal to provide an integrated voice-data signal that is transmitted as a radio signal on a single channel. A receiving station essentially reverses the process to regenerate the voice signal and data signal.

Description

INTEGRATED VOICE AND DATA TRANSMISSION BASED ON BIT

IMPORTANCE RANKING

FIELD OF THE INVENTION

This invention relates to the transmission and receiving of analog and data traffic. Particularly, the invention relates to the integration of a digitized analog signal with a separate data signal to provide an integrated signal. Particularly, the invention relates to the integration of voice and data traffic in radio communication systems. More specifically, the invention relates to the integration of voice and data signals in cellular radio systems.

BACKGROUND OF THE INVENTION Over the years, a number of ways have been proposed to integrate the transmission of voice and data traffic. For example, the spectral characteristics of human speech allow its separation into three bands that have different degrees of usefulness in the reproduction of the speech signal at the receiver: DC to 300 Hz, 300-3200 Hz, and 3200- 4000 Hz. The first and third of these bands are relatively unimportant to the communication of speech.

To exploit this unimportance for integrating the transmission of voice and data traffic, an early method used, for example, a set of filters to shoehorn a low-bit-rate' data channel into either the DC-300 Hz band or the 3200-4000 Hz band, which operated simultaneously with the transmission of speech over the 300-3200 Hz band. More recent proposals suitable for use in a radio communication system such as a cellular telephone system have advocated at least two general approaches: (1) collecting scraps of unused radio frequency ("RF") spectrum (rather than audio spectrum as mentioned above) in one way or another, and using these scraps to overlay a data communication capability; and (2) devoting RF spectrum in an orderly way to providing a set of channels for data communications alongside the set of channels for digitized speech.

Each of these approaches has its advantages and disadvantages, which are well known in the art SUMMARY OF THE INVENTION

A system is set forth for integrating an analog source with a separate data source to provide an integrated signal, transmitting the integrated signal, receiving the integrated signal and regenerating the analog signal and data source. In one embodiment an analog voice signal is converted to a digital signal comprising a plurality of bands. The energy of each band is approximated and compared by means of an importance gauge to a predetermined threshold level in order to identify high energy bands. The information in the high energy bands is compressed and integrated with a separate data signal to provide an integrated voice-data signal that is transmitted as a radio signal on a single channel. A receiving station essentially reverses the process to regenerate the voice signal and data signal.

Accordingly, an object of this invention is to provide a system for integrating a digitized analog signal with a separate data signal to provide an integrated signal.

Another object is to provide a system for integrating a voice signal with a data signal to provide an integrated signal.

Another object is to provide a system for transmitting an integrated signal.

Still another object is to provide a system for receiving an integrated signal.

Another object is to provide a system for regenerating an analog signal and a separate data signal from an integrated signal. A further object is to provide a system for regenerating a voice signal and a separate data signal from an integrated signal.

BRIEF DESCRIPTION OF THE DRAWINGS

The following is a brief description of the drawings that are presented for the purposes of illustrating the invention and not for purposes of limiting the same. FIG. 1 illustrates a block diagram of a transmitter known in the prior art.

FIG. 2 illustrates a block diagram of a frame known in the prior art. FIG. 3 illustrates a block diagram of a receiver known in the prior art. FIG. 4 illustrates a block diagram of a communication system including a transmitter and a receiver known in the prior art. FIG. 5 illustrates a block diagram of a frame known in the prior art. FIG. 6 illustrates a block diagram of the overall functional logic arrangement of a communication system including a transmitter and receiver according to one embodiment of the invention.

FIG. 7 illustrates a block diagram of a transmitter according to one aspect of the invention.

FIG. 8 illustrates a flow chart depicting the logic steps involved in integrating a voice signal with a data signal according to one aspect of the invention.

FIG. 9 illustrates a block diagram of a receiver according to one aspect of the invention. FIG. 10 illustrates a flow chart depicting the logic steps in processing an incoming integrated voice-data signal according to one aspect of the invention.

FIG. 1 1 illustrates a block diagram of a receiver according to one aspect of the invention.

DETAILED DESCRIPTION OF THE INVENTION While the invention will be described in connection with one or more embodiments, it will be understood that the invention is not limited to those embodiments. On the contrary, the invention includes all alternatives, modifications, and equivalents as may be included within the spirit and scope of the appended claims.

Sub-band Speech Coding of the Prior Art The purpose of source coding is to remove redundancy from a waveform that is to be digitized and communicated across a bandwidth-limited channel. One well established source-coding method is sub-band coding. The terms "speech", "voice", and "sound" will hereafter be regarded as equivalent terms.

Referring to FIG. 1 , which depicts a transmitter 5 incorporating a sub-band speech coder known in the prior art, a microphone 10 picks up sound energy in the form of an analog waveform. The analog waveform is sampled digitally at the rate of 8,000 samples per second by an analog to digital converter 20, each sample having perhaps 16 bits of resolution. The resulting outpouring of samples is grouped into a sequence of binary blocks. There are 128 samples per block, 16 bits per sample, for a total of 2048 bits per block. Each block has duration of 16 milliseconds ("ms")

The binary blocks are put through a bank of parallel bandpass filters at 30 to provide a set of band-limited signals. For example, a bank of eight bandpass filters may be employed as depicted in FIG 1 Thus for the audio range of 0-4000 Hz, the first filter can be configured to pass 0-500 Hz, the second filter 500-1000 Hz, ... and the eighth 3500-

4000 Hz. Quadrature mirror filters may be used to cancel sample aliases when the receiver reassembles the bands into a unified signal

To reduce the number of bits sent across a channel, the output from bandpass filters 30 is compressed or re-coded at a lower rate by means of a shared quantizer

40. The output from the quantizer 40 is modulated at 50 in order to generate an output signal 60

Within the quantizer, a total of 512 bits rather than 2048 bits are dedicated to the coding of each 16-millisecond block A duration of 16 ms and a block size of 512 bits ensures an output rate of 32 kilobits per second ("Kps") FIG 2 shows how the block of

512 bits at 55 might be packaged and transmitted to the receiver 51, see FIG 3

The spectral characteristics of human speech allow its separation into a plurality of bands that have different degrees of usefulness in the reproduction of the speech signal at the receiver The exemplary coder of FIG I employs eight bands, which are encoded unequally by the quantizer The re-coding is deliberately unequal to reflect the relative importance of the various outputs from the bandpass filter bank 30 Thus, the 512 bits might be distributed as depicted in Table 1 and FIG. 2

For example, the contribution of band 4 is encoded with 7 bits per sample whereas the contribution of band 8 is encoded with only 2 bits per sample, see Table 1 The unequal bit coding exploits the relative importance between the various outputs from the filter bank 30 and helps to ensure that the voice signal is encoded faithfully within the constraint of 512 bits per block

There are some problems inherent in the transmitter of FIG. 1. For example, the allocation of bits, albeit unequal, fails to take account of real time changes in the nature of the analog signal picked up by the microphone 10. The allocation of bits among the bands remains fixed, as shown in Table I and FIG. 2, even if the voice signal varies with time from voiced to unvoiced, high pitch to low pitch and so on.

Referring to FIG. 3, which illustrates a receiver 5' known in the prior art, the output from transmitter 5 of FIG. 1 is received as an incoming signal 60'. The signal 60' is demodulated at 50' and directed to a bank of parallel bandpass; filters at 30'. A dequantizer 401 followed by a serializer 90 processes the output from the filter bank 30'. The output from the serializer 90 is directed through a digital/analog ("DIX') converter 20' in order to reproduce an analog waveform that is directed to a speaker 10'.

The receiver 5' must know a priori how the transmitter 5 apportioned bits to each band in order to ensure a faithful reproduction of the voice signal at speaker 10' Consequently inefficiencies in the transmission of the voice signal from the transmitter 5 of FIG. 1 are carried over to the receiver 5' of FIG. 3.

FIG. 4, which depicts a communication system known in the prior art, includes a transmitter 5 and receiving apparatus 5'. The transmitter 5 incorporates a sub- band coder which adjusts the allocation of bits to each band as the nature of the speech signal varies with time from voiced to unvoiced, high pitch to low pitch and so on. The energy of each band from the filter bank 30 is approximated and communicated to a bit allocation algorithm 120. The bit allocation algorithm 120 allocates proportionally more bits to bands where substantial energy is detected. Since the receiver 5' of FIG. 4 can no longer rely on a priori knowledge of the transmitter, information on the allocation of bits by the bit allocation algorithm 120 is stored in a bit allocation information field 130 (see FIG. 5). The bit allocation field 130 is incorporated in each frame 140. Each frame 140 may be formatted as shown in FIG. 5.

A complimentary bit allocation algorithm 120' (FIG. 4) reads each bit allocation field 130 in each received frame 140 in the incoming signal 60'. The bit allocation algorithm 12' configures an inverse dequantizer 40' in response to the contents in each bit allocation field 130 in order to correctly reconstruct the voice signal. The reconstructed voice signal is serialized at 90 and converted into an analog signal by the D/A converter at

20' and output through speaker 10'.

The bit allocation field 130 may contain a literal map of the bits that Mow in the frame 140. Alternatively, the bit allocation field 130 may contain a coarse version of the information upon which the transmitter 5 relied when allocating the bits to be sent, so that the receiver 5' and the transmitter 5 might have the same information and act upon it in the same way thereby eliminating the need for a literal map. For example, a coarse approximation of the energy in each sub-band might be sent in the bit allocation field 130. Alternatively, the bit allocation field 130 may contain a coarse spectral profile like that described in U.S. Patent Number 4,644, 108. Coders in the prior art often waste bandwidth. The unimportant bits are often devoted to increasing the coding fidelity of the voice signal regardless of actual need. The terms "unimportant bite', "expendable bits" and "extra bite' will hereafter be regarded as equivalent terms. For example, where the output from the filter bank 30' or the speaker 10' is only required to meet a modest output specification, devoting the unimportant bits to improving coding fidelity is essentially wasteful. Also, when a speech spectrum is very narrow or when the speech spectrum is essentially noise during the pronunciation of a fricative then devoting all of the bits to encoding the speech amounts to wasted bandwidth. Thus, there is a need to avoid wasting bandwidth.

Thus far all of the available bits (512 bits per block in the examples given thus far) are devoted to coding and reproducing the voice signal. Bands in the voice signal with low energy may serve to free up bits ("unimportant bits") that are then devoted to coding higher energy bands in the voice signal.

Speech Coding in the Present Invention

In the present invention the unimportant bits are used primarily to code at least one separate data signal thus permitting an analog signal that has been sampled or digitized and at least one data signal to be communicated in an integrated fashion to a receiver.

Although the invention is described in terms of speech or voice signals, the invention applies as well to any digitized analog signal, including but not limited to image, video, and FAX signals, as well as to digitized music including music encoded according to the NT3 standard, and also to high-bit rate signals in a native digital mode subject to compression such as 16-bit PCM signals. The type of source coder used in the invention is not critical. The type of source coder may include, but are not limited to: a sub-band coder, a vocoder, a transform coder, and a linear-predictive coder. An important aspect of the invention is the integration of a digitized voice signal and at least one data signal by means of importance bit ranking.

The present invention employs bit-importance ranking to identify extra bits in the voice signal, which extra bits are used to carry a separate data signal. In one aspect of the invention, an analog voice signal is digitized to provide a voice bit stream. The importance of each bit in the voice bit strewn is determined against a predetermined threshold in order to identify each voice bit that is expendable verses each voice bit that is important. Each expendable bit is replaced with a data bit to provide an integrated voice-data bit strewn. The integrated voice-bit stream is modulated to provide a modulated signal. The modulated signal is transmitted to a receiver. The integrated voice-data bit stream incorporates at least one bit allocation field to enable the receiver to separate the voice signal and data signal from the integrated voice-data signal.

FIG. 6 depicts a communication system 200, according to one embodiment of the invention, including a transmitter 210 and receiver 210^*. A first source 215, here an analog voice source, is encoded by a source coder 220 of the present invention to provide a voice bit stream. The importance of each bit in the voice bit stream is analyzed by an importance gauge 230 such as a comparator. The terms "importance gauge" and "comparator" will hereafter be regarded as equivalent terms. The importance gauge 230 compares the importance of each bit against a predetermined threshold. Those bits in the voice bit stream that fall below the threshold are deemed expendable. The importance gauge 230 may be absolute, as just described, meaning that the importance gauge 230 judges the importance of the bits of the digitized analog signal with regard to a threshold but without regard to the nature of the data signal. More generally, the importance gauge may be relative, meaning that the importance gauge 230 compares the importance of the bits of the digitized analog signal with the importance of the data signal. For example, if a buffer holding the data signal is approaching its maximum capacity, then the data signal may be given priority over the least important bits of the digitized analog signal irrespective of a threshold. It should be understood that in this aspect of the invention the least important bits of the digitized analog signal qualify as expendable bits. In other cases, the data signal may have enhanced priority when it has been held without opportunity for transmission past an expiration time. In yet another example, the data signal may have enhanced priority relative to the bits of the digitized speech signal when the data signal has an urgent purpose such as the communication of a network management alarm or network congestion relief instruction. The comparator 230 configures a multiplexor 240 of the present invention, wherein the output from the source coder 220 and the signal source 225 are integrated to provide an integrated voice-data signal. The integration takes the form of replacing expendable bits with bits from the second signal source 225. The integrated voice-data signal is modulated by a modulator 250 of the present invention to provide an outgoing integrated signal 260.

The modulated output signal 260 is received as an incoming integrated signal 2601 with respect to the receiver 210'. The signal 260' is demodulated at 250' and the comparator 230 determines the importance of each bit. In one embodiment, the comparator 230 examines the bit-allocation information 130 (see FIG. 5) in order to determine the importance of each bit in the demodulated signal. The comparator 230 configures a demultiplexer 240'. The demultiplexer 240' separates the demodulated signal into a voice bit stream and a data bit stream. The voice bit stream is decoded by a source decoder 220' and forwarded to sink #1 at 215'. The data bit stream is directed to sink #2 at 225'.

Referring to FIG. 7, which shows a transmitter 210 according to another aspect of the invention, a speech source 215 is digitized by means of an A/D converter 218 and separated into bands by a filter bank 222. A bit allocation gauge 280 approximates the energy of each band. A shared quantizer 235 processes the bands. The bit allocation gauge 280 communicates with a bit allocation algorithm 290, an importance gauge 230 and a multiplexor 240. The bit allocation algorithm 290 configures the shared quantizer 235. The multiplexor 240 receives the output of the shared quantizer 235 and is configured by data received from the importance gauge 230 and the bit allocation gauge 280. When configured to do so, the multiplexor 240 integrates the output of the shared-quantizer 235 with digital data from a digital source 225. The output of the multiplexor 240 is modulated at 250 to provide an outgoing integrated signal 260.

In more detail, a voice signal at 215 is digitized at 218 to provide a digitized voice signal. The digitized voice signal is passed through a set of parallel band-pass filters

222 to provide a set of bands. A bit allocation gauge 280 approximates the energy of each band. The approximate energy of each band is forwarded to the importance gauge 230. The importance gauge 230 compares the energy of each band against a predetermined threshold and thereby identifies bits that are important versus bits that are expendable. Alternatively, the energy of each band is compared against a universal threshold. When the energy for a band is above the threshold for that band, bits are allocated conventionally to the quantizer 235 according to a bit allocation algorithm at 290. When the energy for every band in a block is above the threshold, all available bits ("M") are toward the conventional encoding of the voice signal from the analog speech source 215 (the first source). When the energy all the bands in a block are above the threshold, the multiplexor 240 is appropriately configured.

Referring to FIG. 8, which depicts a flow chart according to a further aspect of the present invention, the analog speech source 215 is sampled and digitized by the A/D converter (not shown) and separated into bands at 218. The energy of each band is approximated at 280. The energy of the lowest energy band in each block is compared to the threshold at 230. If it is found, at 270, that the lowest energy band is above the threshold then all the bits (i.e. M bits) representing the block are allocated to the analog signal according to a first bit algorithm at 292.

If it is found, at 270, that the lowest energy band is below the threshold then a second bit algorithm at 300 allocates N bits, which would otherwise have been allocated to the low energy bands, to the data signal from the second source 225. The second bit algorithm then allocates M-N bits at 310 to the high-energy bands. The speech and data signal are encoded at 320, multiplexed and transmitted along with energy estimates or hard map at 330. The energy estimates for each band or hard map may be encoded in the bit allocation field 130 of each frame 140 at 330.

Though FIG. 8 suggests that the allocation of N bits to the data signal and M-N bits to the voice signal occurs in series, it should be understood that the allocation of N bits to the data signal 225 and M-N bits to the voice signal 215 may alternatively be carried out in parallel. In one aspect of the invention if the energy of a band is determined to be below the threshold value, the N-bit-allocation for that band is forced to zero, and bits are allocated to other bands according to a second bit allocation algorithm that allocates M-N bits for encoding the speech signal from the analog source 215 and N bits towards encoding the data signal from the second source 225. Whenever a bit allocation nominally intended for encoding speech is forced to zero, the multiplexor 240 is configured by the comparator 230 to integrate bits from the second source 225 into the nominal voice bit stream in order to provide an integrated voice-data bit stream. Thus, expendable bits in the voice signal are replaced with bits from a separate data signal to provide an integrated voice-data signal. The integrated voice-data signal is modulated at 250 (FIG. 7) to provide an outgoing integrated signal 260.

The importance gauge 230 might determine the importance of the bits in each band by comparing the approximate energy of each band against a predetermined threshold. For example, if the energy of band 2 were low, i.e. below the threshold, then all of the bits in band 2 would be deemed to be unimportant and band 2 would not be encoded. Thus the bits in the 512-bit block within each frame 140 that would normally be allocated to band 2 would be available to encode bits from a second signal source 225. The energy estimates for each band or hard map may be encoded in the bit allocation field 130 of each frame 140, see FIG. 5. In one aspect of the invention the bit allocation field 130 of each frame 140 is read by the receiver 210' (see FIG. 9) and used to configure the de-multiplexor 240' and inverse quantizer 235'. In principle, if the bit allocation field is used to explicitly store a hard map of each 512-bit block in each frame

140 in the integrated voice-data signal 260, then the bit allocation field may be read by the bit allocation algorithm 290' and used to configure the de-multiplexor 240' and inverse quantizer 235' without requiring the functionality of a comparator 230 at the receiver 210', see FIG. 1 1. In essence, the transmitter 210 would have performed the logical steps to integrate the voice and data signals along with transmitting the information necessary to separate the integrated signal 260' at the receiver 210'.

It should be understood that when the bit allocation nominally intended for speech is forced to zero, bits are freed-up. The freed-up bits are not allocated to improve the fidelity of the speech signal but rather, the freed-up bits are allocated by a second bit algorithm, see FIG. 8, to carry bits from a second source 225 such as a data signal.

It should be further understood that the comparator 230 is not restricted to evaluating the importance of bits according to the approximated energy of a band. For example, the comparator 230 may determine the importance of bits according to traffic class. For example, network management or control information may be judged more important than the signal provided by band 8 of a speech coder. Alternatively, network management or control information may be regarded as more important than the lowest- energy bands without regard to a threshold. Alternatively, the comparator 230 may judge the importance of bits according to tariffing considerations. Thus, it should be understood that the comparator might be configured or programmed to judge the importance of bits according to any measurable factor or absolute factor that happens to be of interest at that time.

Referring to FIG. 9, which depicts one aspect of the invention, the receiver 210' receives an incoming integrated signal 260' which is demodulated at 250' to provide a demodulated signal. The bit allocation field 130 in the demodulated signal is analyzed by the comparator 230 and compared against a threshold. If the energy of the band with the lowest energy is above the threshold then the de-multiplexor 240' is appropriately configured and M bits are allocated in the conventional way to the filter bank 222'. M bits corresponding to the voice signal are processed by the inverse quantizer 235', and serialized at 238 to provide a serialized signal. The serialized signal is converted to an analog signal at 218'. The analog signal is output at speaker 215'. In contrast, when the energy of the lowest energy band falls below the threshold, N bits corresponding to the data signal are allocated to source #2 according to a second bit algorithm (see FIG. 10). M-N bits in high-energy bands are allocated to the speech source. The demultiplexer 240' is appropriately configured to demultiplex, the bit stream. N bits are directed to sink #2 at 225', and M-N bits to the filter bank 222'. M-N bits corresponding to the voice signal are processed by the inverse quantizer 235, serialized at

238, converted to an analog signal at 218' and regenerated as a sound output at 215'. In this embodiment of the invention the receiver 210' essentially reverses the process of the transmitter 210 to regenerate the voice signal and data signal.

Referring to FIG. 10, which depicts a flow chart showing the operation of the receiver 210', the incoming integrated signal 260' is demodulated to. provide a demodulated signal. Each bit-allocation field in the demodulated signal is examined at 360 and the lowest energy is compared to the threshold at 230'. If the lowest energy band in a 512-bit block is above the threshold at 270' then all 512 bits are allocated to the speech signal according to a first bit algorithm at 292' and are processed in a conventional manner. If the lowest energy band is below the threshold then N bits are allocated at

370 to source #2 in nominal positions of low-energy bands. M-N bits are allocated to the speech source at 380. The de-multiplexor is configured and de-multiplexes the demodulated signal at 240'. N bits are routed to sink #2 at 390. M-N bits are decoded at 400 and the speech signal is processed in the conventional way at 410. It should be understood that expendable bits are replaced with data bits according to demand. Periods might arise when there are no data bits available to replace the expendable bits. During these periods the information stored in the bit allocation field 130 will enable the receiver 5' to allocate bits to reconstruct the analog signal according to a 1^st bit algorithm (see FIG. 10). While the invention is described above in connection with preferred or illustrative embodiments, these embodiments are not intended to be exhaustive or limiting of the invention. Rather, the invention is intended to cover all alternatives, modifications and equivalents included within its spirit and scope of the invention, as defined by the appended claims. In particular, the invention applies to all kinds of source coders, not just to the sub-band speech coder used herein for purposes of clear description.

Claims

CLAIMSWE CLAIM:

1. A method of integrating an analog signal and a digital signal, for communication over a common channel, comprising the steps of: digitizing the analog signal to provide a bit stream; estimating importance of bits of the bit stream to identify at least one expendable bit; replacing said at least one expendable bit by at least one bit from the digital signal to provide an integrated bit stream; and communicating the integrated bit stream over the common channel.

2. The method according to claim 1, wherein said analog signal is selected from the group comprising of a voice signal, an image signal, a video signal, and a fax signal.

3. The method according to claim 1 further comprising the step of encoding said first digital signal by a source coder

4. The method according to claim 3, wherein said source coder is selected from the group comprising of a sub-band coder, a vocoder, a transform coder, and a linear predictive coder.

5. The method according to claim 1, wherein the estimating step estimates the importance of bits of the bit stream against a threshold.

6. The method according to claim 1 , wherein the bit stream is separated into bands and the energy of each band is approximated and compared by means of an importance gauge against a threshold level in order to identify the at least one expendable bit.

The method according to claim 6, wherein a bit allocation algorithm allocates bits from the digital signal in place of expendable bits to provide said integrated signal.

8. The method of claim 1, wherein the replacing step further comprises replacing expendable bits of the bit stream with high priority bits in the second signal.

9. The method of claim 8, wherein the high priority bits are selected on the basis of expiration time, communication of a network management alarm or a network congestion relief instruction.

10. A method of integrating a first signal and a digital signal, for communication over a common channel, comprising the steps of processing the first signal to provide a bit stream; estimating importance of bits of the bit stream to identify at least one expendable bit; replacing said at least one expendable bit by at least one bit from the digital signal to provide an integrated bit stream; and communicating the integrated bit stream over the common channel.

11. The method according to claim 10, wherein said first signal is selected from the group comprising of a voice signal, an image signal, a video signal, and a fax

12. The method according to claim 10, wherein the estimating step prioritizes importance of bits of the bits stream relative to the second signal to identify the at least one expendable bit.

13. A method of processing an integrated signal received over a communication channel, comprising the steps of:

(a) receiving the integrated signal, said integrated signal comprises at least one bit allocation field; (b) analyzing said at least one bit allocation field in order to identify bits of the integrated signal that correspond to a first signal and bits that correspond to a second signal;

(c) separating bits of the first signal and bits of the second signal from the integrated signal; and

(d) outputting said first signal and said second signal.

14. The method according to claim 13 , wherein the bit allocation field comprises at least one approximated energy value, and said analyzing step further comprises comparing approximated energy values against a threshold to identify bits of the integrated signal that correspond to the first signal and bits that correspond to the second signal.

15. The method according to claim 13 , wherein the bit allocation fields store a hard map of bits of the integrated signal, and said analyzing step further comprises analyzing said hard map to identify bits of the integrated signal that correspond to the first signal and bits that correspond to the second signal.

16. The method according to claim 13, wherein the outputting step further comprises the steps of (e) filtering said first signal to provide a -plurality of bands, (f) dequantizing said plurality of bands to provide a dequantized signal, (g) serializing the dequantized signal to provide a serialized signal, and (h) converting said serialized signal to an analog signal.

17. A communication system for generating an integrated signal, comprising: means for digitizing an analog signal to provide a bit stream; means for estimating importance of bits of the bit stream to identify at least one expendable bit; means for replacing said at least one expendable bit by at least one bit from a digital signal to provide an integrated bit stream; and means for communicating the integrated bit stream over a common channel.

18. The communication system of claim 17, wherein said analog signal is selected from the group comprising of a voice signal an image signal, a video signal, and a fax signal.

19. A communication system for generating an integrated voice-data signal and receiving said integrated voice-data signal, comprising: means to convert a voice signal into a first digital signal; means to determine the importance of each bit in first digital signal against a predetermined threshold and thereby identify bits that are expendable; means to replace, said expendable bits with bits from a second digital signal to provide an integrated voice-data signal; means to modulate said integrated voice-data signal to provide an outgoing integrated voice-data signal; means to receive said outgoing signal as an incoming integrated voice-data signal; and means to separate said incoming signal to provide a separate voice signal and data signal.