ES2231899T3

Method and apparatus for compressing and transmitting high speed data

Abstract

TWO RELATED VOICE BAND COMPRESSION TECHNIQUES ARE USED TO ALLOW AN RF TELECOMMUNICATION SYSTEM TO ALLOW THE DATA SIGNALS FROM HIGH SPEED VOICE BAND MODEMS AND FAX MACHINES. THROUGH A HIGH-SPEED CODE CODE, THE TELECOMMUNICATION SYSTEM IS PASSED TO TRANSMIT THE TRANSMISSIONS OF THE VOICE BAND MODEM AND FAX MACHINES AT SPEEDS UP TO 9.6 KB / S. A ULTRA HIGH SPEED CODE-DECODER SUPPORTS THE TRANSMISSIONS OF THE VOICE BAND MODEM AND FAX MACHINES UP TO 14.4 KB / S. THE HIGH SPEED ENCODER-DECODER USES THREE RF SEGMENTS OF 16 PHASES OR FOUR 8-PHASE RF SEGMENTS, AND THE ULTRA HIGH-SPEED ENCODER-DECODER WORKS WITH FOUR 16-PHASE RF SEGMENTS. AS THESE CODE-DECODERS TRANSMIT INFORMATION ON VARIOUS RF SEGMENTS THAT MAY BE CONTIGATED, A DYNAMIC ASSIGNMENT OF THE SEGMENTS IS CARRIED OUT INSIDE THE RF COMMUNICATION CHANNELS. THROUGH THE CHARACTERISTICS DYNAMIC ASSIGNMENT OF TIME / BANDWIDTH SEGMENTS THE DATA TRANSMISSION IS DETECTED AND CONTROLLED AND A DATA CHANNEL IS ESTABLISHED FROM THE NUMBER REQUIRED OF SEGMENTS.

ES2231899T3, drawing sheet 1 — Sheet 1 of 16

Term

Term ended

Projected expiry passed 4 November 2017, 8.9 years ago.

18 claims: 2 independent, 16 dependent

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ES 2 231 899 T3 ES 2 231 899 T3 CLAIMS REIVINDICACIONES 1. A telecommunications apparatus for receiving a plurality of telephone signals and for transmitting each of the telephone signals over a respective communication channel, wherein each communication channel is formed on at least one transmitting radio frequency (RF) carrier, each RF carrier having a plurality of information segments and at least one of the information segments being assigned to one of the telephone signals so that said one of the telephone signals is modulated onto the RF carrier, the apparatus comprising:1. Un aparato de telecomunicaciones para recibir una pluralidad de señales telefónicas y para transmitir cada una de las señales telefónicas por un canal de comunicación respectivo, en el que cada canal de comunicación está formado sobre al menos una portadora de radiofrecuencia (RF) de transmisión, teniendo cada portadora de RF una pluralidad de segmentos de información y estando asignado al menos uno de los segmentos de información a una de las señales telefónicas de modo que dicha una de las señales telefónicas se module sobre la portadora de RF, comprendiendo el aparato: encoding means (210, 220, 230, 240, 250) for encoding the telephone signals in order to generate encoded signals, and means for modulating the encoded signal in the communication channel, characterized in that the apparatus further comprises detector means ( 202) to receive and monitor each of the telephone signals to detect a data signal in one of the telephone signals;medios de codificación (210, 220, 230, 240, 250) para codificar las señales telefónicas a fin de generar señales codificadas, y medios para modular la señal codificada en el canal de comunicación, caracterizado porque el aparato comprende, además, medios detectores (202) para recibir y supervisar cada una de las señales telefónicas a fin de detectar una señal de datos en una de las señales telefónicas;control means (201) for verifying an allocation status of some of the information segments in response to the detection of the data signal and for locating a predetermined number of unallocated sequential information segments for a predetermined bandwidth, indicating the assignment status if each information segment is unassigned or assigned to a respective one of the telephone signals;medios de control (201) para verificar un estado de asignación de algunos de los segmentos de información en respuesta a la detección de la señal de datos y para localizar un número predeterminado de segmentos de información secuenciales no asignados para un ancho de banda predeterminado, indicando el estado de asignación si cada segmento de información está sin asignar o se encuentra asignado a una respectiva de las señales telefónicas;channel forming means (260) for forming the communication channel from the unassigned sequential information segments in accordance with a dynamic information segment allocation based on the number of unassigned sequential information segments. medios de formación de canal (260) para formar el canal de comunicación a partir de los segmentos de información secuenciales no asignados de acuerdo con una asignación dinámica de segmentos de información basada en el número de segmentos de información secuenciales no asignados.
10
A communications method for receiving a plurality of telephone signals and for transmitting each of the telephone signals over a respective communication channel in a telecommunications system, wherein each communication channel is formed on at least one radio frequency carrier (RF ) of transmission, each RF carrier having a plurality of information segments and at least one of the information segments being assigned to one of the telephone signals so that said one of the telephone signals is modulated on the RF carrier, the method comprising the steps from:10. Un método de comunicaciones para recibir una pluralidad de señales telefónicas y para transmitir cada una de las señales telefónicas por un canal de comunicación respectivo en un sistema de telecomunicaciones, en el que cada canal de comunicación está formado sobre al menos una portadora de radiofrecuencia (RF) de transmisión, teniendo cada portadora de RF una pluralidad de segmentos de información y estando asignado al menos uno de los segmentos de información a una de las señales telefónicas de modo que dicha una de las señales telefónicas sea modulada sobre la portadora de RF, comprendiendo el método los pasos de: a) receiving and monitoring each of the telephone signals to detect a data signal in one of the telephone signals;a) recibir y supervisar cada una de las señales telefónicas para detectar una señal de datos en una de las señales telefónicas;b) codificar la señal de datos para generar una señal codificada;b) encoding the data signal to generate an encoded signal;c) verificar un estado de asignación de algunos de los segmentos de información en respuesta a la detección de la señal de datos, indicando el estado de asignación si cada portadora y cada segmento de información están sin asignar o han sido asignados a otra de las señales telefónicas;c) verifying an assignment status of some of the information segments in response to the detection of the data signal, indicating the assignment status if each carrier and each information segment are unassigned or have been assigned to another of the signals telephone;d) localizar un número predeterminado de segmentos de información secuenciales no asignados;d) locating a predetermined number of unassigned sequential information segments;e) formar el canal de comunicación a partir de los segmentos de información secuenciales no asignados de acuerdo con una asignación dinámica de segmentos de información basada en el número de segmentos de información secuenciales no asignados;y e) forming the communication channel from the unassigned sequential information segments according to a dynamic information segment assignment based on the number of unassigned sequential information segments;Y f) modificar la señal codificada en el canal de comunicación. f) modify the encoded signal in the communication channel.

Independent claims2

351 paragraphs in 20 sections, as filed

ES 2 231 899 T3

DESCRIPTION

Method and apparatus for compressing and transmitting data at high speed.

Field of the invention

This invention relates to a communication system and more particularly to signal processing techniques for the compression of high speed data communication signals in order to obtain improved transmission performance and increased communication system capacity. .

Background of the invention

Telecommunication systems are well known in the art and today's telephone systems employ various multiplexing techniques to transmit telephone signals from many users over a single transmission line, such as wire or fiber optic cable. Most of these “hard-wired” systems employ a form of time division multiplexing (TDM) in which multiple channels are transmitted sequentially at speeds higher than the information rate of the channels.

Typical telephone multiplexing requires sampling the telephone signal and transmitting the samples at a frequency much higher than the frequency of the telephone signal. To this end, current systems digitally sample and encode the telephone signal, multiplex and transmit the signal, and then receive, demultiplex, and decode the signal. One such sampling and coding system is a pulse code modulation (PCM) in which analog signals from the voice band are sampled at a rate of 8 kilosamples per second, each sample being represented by 8 bits. Consequently, the voiceband signal is converted to a 64 kilobits per second (kb / s) digital signal.

Another form of telecommunication system is the radiotelephone system. Radiotelephone systems use a group of selected radio frequencies (RF) to carry telephone communication signals between two or more locations and typically employ a form of frequency division multiple access (FDMA). These radio systems, called wireless communication systems, are used, for example, in rural places to provide local telephone service or in mobile units to provide mobile communication service.

One category of RF communication systems employs TDM to allow user access to multiple time slots of information modulated on the RF carrier. If many users compete for a small group of information time slots, the system is called time division multiple access (TDMA). In order to allow TDMA of the RF communication channels FDMA a method, called FDMA / TDMA and described in the patent US 4,675,863 has been used to increase the capacity of the RF communication systems. However, RF communication systems are often still limited in their capabilities when compared to hard-wired or fiber optic communication systems.

Consequently, to further increase capacity, signal compression techniques have been used to deduce the bandwidth required for the transmission of a telephone signal over an RF channel. Typical techniques used for speech signals are subband coding, adaptive differential pulse code modulation (ADPCM), and residual linear predictive coding (RELP). RELP or similar voice compression algorithms allow a 64 kilobits per second (kb / s) sampled and quantized voice signal to be transmitted over the RF channel as a low bit rate signal (for example, 14.6 kb / s or less). The receiver reconstructs the 64 kb / s speech signal from the low bit rate signal and the listener perceives little or no loss in signal quality.

The underlying speech compression method, including RELP, is an encoding and decoding algorithm that takes advantage of known characteristics of speech signals. One type of RELP method adopts certain characteristics of the harmonics of the human voice. However, today, a large part of the communication signals within a telephone network are non-voice data communications, such as facsimile data (FAX) or voice band modem data. Unfortunately, voice compression algorithms are not particularly compatible with these data communication signals because the data signals do not have the characteristics of voice signals.

Consequently, some RF communication systems monitor the telephone signal for the presence of a data communication signal. Typically, data signals representing modem or FAX data signals have been detected in voice band up to 2.4 kb / s (low speed data) and these signals have been provided with a specialized compression algorithm. The receiver reconstructs the data signal without reducing the transmission data rate. This system and method are described in, for example, US Patent 4,974,099.

However, today's telephone data signals are more typically 9.6 kb / s (high-speed data) or higher (ultra-high-speed data, such as 14.4 kb / s or 28.8 kb / s). other higher or lower rates), and the present compression techniques do not satisfactorily compress these higher data rates. Compression of these higher data rates and especially multiple encodings of these higher data rates cause a degradation in the quality of the modem or FAX signals, and the modem or host machine

ES 2 231 899 T3

FAX will frequently reduce the data transmission speed when the signals are passed through an RF communication system.

Patent US 5 446 739 describes a radio communication system in which audio and data transmissions can be carried out. In the case of data transmissions, two time slots are allocated not adjacently on different frequencies or adjacently on the same frequency. Segment allocation is done in response to a request from the cell station for a particular number of segments.

WO 96/27975 describes a mobile communication system for transmitting facsimile signals over a GSM interface. In order to establish an appropriate communication transmission rate, the maximum data rate is obtained from the mobile station and a corresponding number of channels is assigned to the call. The data transmission speed is then adjusted, once the call has been established, if the resulting quality is too low. This change in data rate is achieved by decreasing the ratio of the convolutional code.

Summary of the invention

The present invention provides a telecommunications apparatus according to claim 1 and a method of receiving a plurality of telephone signals according to claim 10. Other preferred aspects of the invention are provided in accordance with the dependent claims.

A telecommunication system receives a group of telephone signals, including data signals, each of which has a form of encoding, and transmits the telephone signals on at least one radio frequency (RF) carrier. Each RF carrier has a group of information segments and each telephone signal is assigned to at least one information segment so that the telephone signal is modulated on the RF carrier. The system includes a method for monitoring and identifying the data signals and for compressing each data signal to reduce the required transmission bandwidth of the data signal.

Brief description of the drawings

The invention will be best understood from the following detailed description when it is read in conjunction with the accompanying drawings, in which:

Figure 1 is a block diagram of a wireless communication system.

Figure 2 is a high-level block diagram of the implementation of the compression system of the present invention, including the dynamic bandwidth allocation feature and the high-speed and ultra-high-speed data codecs.

Figure 3A is a high-level flow chart illustrating the detection and selection of high-speed data encoding types and the determination and allocation of radio channel segments in accordance with an exemplary embodiment of the present invention.

Figure 3B is a high-level flow chart showing the channel allocation process performed by the channel forming processor upon request for a high-speed data channel in accordance with one embodiment of the present invention.

Figure 4A is a graph showing the characteristics of an A-law quantizer.

Figure 4B is a graph showing PCM quantization signal-to-noise performance as a function of uniform quantization.

Figure 4C illustrates the compression method by mapping signal samples from one quantization to another.

Figure 5A is a high-level block diagram of the high-speed data encoder in accordance with an exemplary embodiment of the present invention.

Figure 5B illustrates a high speed data encoder transmission encoding process in accordance with an exemplary embodiment of the present invention.

Figure 6A is a high-level block diagram of the high-speed data decoder in accordance with an exemplary embodiment of the present invention.

Figure 6B illustrates a high speed data decoder transmission decoding process in accordance with an exemplary embodiment of the present invention.

Figure 7A is a high-level block diagram of the ultra-high-speed data encoder in accordance with an exemplary embodiment of the present invention.

ES 2 231 899 T3

FIG. 7B illustrates an ultra-high-speed data encoder transmission encoding process in accordance with an exemplary embodiment of the present invention.

Figure 8A is a high-level block diagram of the ultra-high-speed data decoder in accordance with an exemplary embodiment of the present invention.

FIG. 8B illustrates an ultra-high-speed data decoder transmission decoding process in accordance with an exemplary embodiment of the present invention.

Figure 9 is a high-level flow chart illustrating an ultra-high-speed quantization algorithm used to map quantized PCM samples to compressed quantized samples in accordance with an exemplary embodiment of the present invention.

Overview

A telecommunications apparatus and method receive telephone signals and modulate each of the telephone signals onto a respective transmitting radio frequency (RF) carrier. Each transmitting RF carrier has a predetermined number of information segments and each telephone signal is assigned to at least one information segment so that the telephone signal is modulated onto the RF carrier. The telecommunications apparatus and method includes a detector for receiving and monitoring each of the telephone signals to detect a data signal contained in one of the telephone signals; and an encoder for encoding the data signal in the form of a compressed encoded signal. The apparatus and method also include a controller that verifies an allocation status of each information segment when the data signal is detected, and locates a predetermined number of unallocated (but not necessarily contiguous) sequential information segments for a width of The predetermined band required to transmit the compressed encoded signal. Assignment status indicates whether each information segment is unassigned or assigned to other telephone signals. The apparatus and method also include a method of forming a telecommunication channel from the unassigned localized sequential information segments, and a method of modulating the encoded signal on the telecommunication channel.

In accordance with one aspect of the present invention, a high speed data compression transmission system transmits a high speed data signal over a telecommunication channel in the form of a compressed encoded signal. The high speed data signal is received as at least one data signal sample block and the system includes a high speed data encoder and a high speed data decoder. The high speed data encoder includes 1) a receiver for the data signal blocks each containing at least one data signal sample representative of a peak amplitude; 2) a calculator for calculating a gain value of the data signal block that is proportional to the value of the peak amplitude; and 3) a quantizer selector that selects a quantizer corresponding to the gain value.

The quantizer has a plurality of quantization level values having a predetermined (eg, uniform) spacing, which are determined from the gain value, and the selected quantizer quantizes each data sample in the data signal block. in the form of a compressed data sample. The gain value and the plurality of compressed data samples constitute the compressed encoded signal. The high speed data compression transmission system includes a transmitter for transmitting the compressed encoded signal over the telecommunication channel and a receiver for receiving the signal from the telecommunication channel.

The high-speed data decoder of the high-speed data compression transmission system includes 1) a receiver for the compressed data samples and the corresponding gain value; and 2) an inverse quantizer selector for selecting, based on the gain value, a uniform inverse quantizer having a plurality of uniformly spaced output values that are determined from the gain value. The inverse quantizer processes each of the compressed data samples based on the gain value to provide a sample block of the reconstructed data signal.

In accordance with another aspect of the present invention, an ultra-high-speed data compression transmission system transmits an ultra-high-speed data signal over a telecommunication channel. The ultra-high-speed data signal is received as at least one block of samples of the data signal having a first quantization, and the system includes an ultra-high-speed data encoder and an ultra-high-speed data decoder. The ultra high speed data encoder includes 1) a receiver for the data signal block containing at least one data signal sample having a peak amplitude; 2) a calculator for calculating a data signal block gain value that is proportional to the peak amplitude; and 3) a quantizer selector for selecting a new set of quantizer levels corresponding to the gain value of the sample block, and wherein each level of the new set of quantizer levels are selected levels from the first quantization; and 4) a quantizer level mapping processor that maps the signal sample value to a compressed level value for each signal sample value based on a relationship between the set of levels of the first quantization and the new one. set of quantizer levels.

ES 2 231 899 T3

The gain value and the compressed data samples constitute a coded signal. The system also includes a transmitter for transmitting the coded signal on the telecommunication channel, and a receiver for receiving the coded signal from the telecommunication channel. An example of an embodiment is described below with reference to a telecommunication channel of a wireless communication system. However, the present invention is not limited to wireless or other types of RF carrier communication. On the contrary, the present invention can also be used with telecommunication channels of wired communication systems to increase their capacity.

The ultra-high-speed data decoder of the ultra-high-speed compression transmission system includes 1) a receiver for the compressed data samples and the corresponding gain value; and 2) an inverse quantizer selector for selecting, based on the corresponding gain value, an inverse quantizer having output values that are determined from the gain value and a corresponding new set of quantizer levels. The inverse quantizer processes each of the compressed data samples based on the gain value to provide a block of reconstructed samples of the data signal.

In accordance with another aspect of the present invention, an ultra-high-speed data quantization method maps from a first plurality of quantized signal samples, each signal sample having a corresponding quantized amplitude value and at least one signal sample having a peak quantized amplitude value, up to a second plurality of quantized compressed samples and a gain value. The method includes 1) examining each amplitude to determine a peak amplitude value and adjusting the gain value corresponding to the peak amplitude value; and defining for the first plurality of quantized signal samples a predetermined number of successive segments, each segment having a plurality of quantized level values. The quantized level values for each successive segment are related to the gain value, and a first segment of the predetermined number of successive segments corresponds to the peak amplitude of the plurality of signal samples.

The quantization method further includes mapping each of the quantized signal samples in the form of quantized compressed samples 1) retaining for each of the quantized signal values selected values from the plurality of quantized level values for each segment up to a zero value level is found, and 2) setting a sign value to a negative value to indicate a negative value amplitude. Detailed description of the invention

The data compression system

Figure 1 is a diagram of a wireless telecommunication system in which the high speed data compression features of the present invention can be implemented. As shown, the radio telecommunications system includes a base station 11 and a group of subscriber units 10. The base station 11 communicates simultaneously with the subscriber units 10 by broadcasting and receiving defined communication channels over a range of preselected radio frequencies. Base station 11 may also interface with local telephone equipment at central telecommunications office 12.

A typical radio telecommunications system (for example, the SLS-104, manufactured by InterDigital Communications Corporation, King of Prussia, Pennsylvania) uses 24 default forward channels (base station to subscriber unit) and 24 default reverse channels (base station to subscriber unit). base station subscriber) within the 300-500 megahertz (MHz) spectral region. The communication from the base station to the subscriber unit is provided by pairs of communication channels (forward and reverse) modulated on frequencies within this spectral region: In a typical system the base station 11 communicates simultaneously on these 24 pairs channels. The 24 channels can occupy, for example, 2 MHz frequency bands. The 2 MHz frequency band can support more channels, for example 80 channels, using 25 kHz channel spacing. In one embodiment of the system the base station 11 can transmit to a subscriber on the inner frequency of a pair, and the subscriber unit 10 can transmit to the base station on the higher frequency pair. One such system is described in US Patent No. 4,675,863, issued June 23, 1987 to Paneth et al. and titled SUBSCRIBER RF TELEPHONE SYSTEM TO PROVIDE MULTIPLE VOICE AND / OR DATA SIGNALS SIMULTANEOUSLY BY SINGLE OR A PLURALITY OF RF CHANNELS.

To increase communication capacity, time division multiple access techniques are used on each carrier frequency. In one example of the system, each frequency of the channel pair is divided into four time slots such that the base station 11 simultaneously communicates with up to four subscriber units 10 on a carrier frequency. Consequently, the base station, using 24 pairs of channels, can allow telephone signals to be modulated into 95 channels and can use one channel for control and other general functions.

One aspect of increasing capacity in this way is to compress the telecommunication channels to be transmitted on the RF communication channel (or wired channel). For speech, as described above, speech coding techniques such as RELP can be used. Also, low-speed data compression and low-speed facsimile data compression techniques, such as those described

ES 2 231 899 T3 in US Patent 4,974,099 to Lin et al., Entitled COMMUNICATION SIGNAL COMPRESSION SYSTEM AND METHOD.

In the above-described system three voice band encoders, RELP, low speed data and low speed FAX, compress 64 kb / s PCM signals to give a 14.5 kb / s signal. At 14.5 kb / s, these three encoders can operate within a single 16-phase RF segment or a double-wide 4-phase RF segment. The RELP encoder is used for voice, the low-speed data encoder is used to pass a plurality of voice band modem transmissions at speeds up to 2400 BPS, and the low-speed FAX encoder is used to pass transmissions. Group 3 FAX at 2400 BPS. Each transmitting encoder has a corresponding decoder within a receiver, which can be assigned, for example, via the system control channel.

To enable the telecommunications system to accommodate voice band modems and high speed FAX machines the two related voice band compression techniques of the present invention are employed. The encoders and decoders (codecs), referred to as the high-speed codec and the ultra-high-speed codec, achieve better compressed data transmission performance than low-speed data and FAX encoders, using less compression and therefore providing more bandwidth to the data signal.

The high-speed codec enables the telecommunications system to pass voice band modem and FAX transmissions at up to 9.6 kb / s. The ultra-high-speed codec supports voiceband modem and FAX transmissions of up to 14.4 kb / s and more. The high-speed codec works using three 16-phase RF segments or four 8-phase RF segments. The ultra-high-speed codec works using four 16-phase RF segments. Preferably, high-speed data and ultra-high-speed data compression algorithms pass a representation of an analog voice band waveform over a digital channel with restricted data rates, while minimizing detrimental distortion.

Since these codecs use multiple RF segments, a dynamic reassignment of the segments is required within the RF communication channels. The dynamic time slot / bandwidth allocation feature of the present invention detects and monitors data transmission and forms a data channel from the required number of segments, but if the required number of segments is not available, it is assigns the low speed data or low speed FAX encoder to the call. Such allocation methods are described, for example, in US Patent No. 4,785,450, issued November 15, 1988 to DR Bolgiano et al. and entitled APPARATUS AND METHOD TO OBTAIN FREQUENCY AGILITY IN DIGITAL COMMUNICATION SYSTEMS.

Figure 2 is a high-level block diagram of the implementation of the compression system of the present invention, including the dynamic time slot / bandwidth allocation feature, and the high-speed and ultra-high-speed data codecs, for high-speed data compression of the exemplary embodiment of a wireless telecommunication system. The system includes: a compression selector processor (CSP) 200 that includes a control unit 201 and a monitor section 202; a channel shaping processor 260; and the RELP 210 compression encoders / decoders (CODECs), low speed data 220, low speed FAX 230, high speed data 240 and ultra high speed data 250.

The CSP 200 receives the telephone signal from the local telephone exchange 270 and is a digital processor designed to implement telephone signal supervision to identify specific types of data signals by their respective modem answer tones and initiate the setting of the telephone signal. communication channel. In another embodiment using subscriber-to-subscriber communications, the CSP 200 may receive the telephone signal from other local sources. The monitor section 202 of the CSP 200 informs the control unit 201 of the presence of the data signal. The control unit 201 is responsible for implementing the external formation of an RF communication channel, as well as assigning a type of compression CODEC 210, 220, 230, 240 and 250.

Channel shaping processor 260 receives a request for a transmission channel from CSP 200 and allocates an available RF communication segment to a telephone signal. Channel shaping processor 260 maintains current system channel assignment information in memory (not shown) to determine which time slots are not currently used for other telephone signals. As is known in TDMA systems, each channel time slot is formed with a guard time which is a short signal period used to initialize the receiver before data is sent. In the presence of data signals that require more than one RF time slot, channel shaping processor 260 forms the channel from a predetermined number of time slots, and if the predetermined number of time slots is contiguous, only one guard time is used.

The channel-forming processor 260 of an exemplary embodiment of the invention may be a radio processor unit (RPU) of a network base station. The RPU may be responsible for storing channel time slice assignments and assigning channel time slices for the entire system of Figure 1.

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The RELP CODEC 210 implements the compression coding (and decoding) algorithms for speech signals. The 220 Low Speed Data CODEC and 230 Low Speed FAX CODEC, 240 High Speed Data CODEC and 250 Ultra High Speed Data CODEC implement the respective data compression algorithms for voice band data from the identified type.

In general, the CSP 200 and CODECs 210, 220, 230, 240, and 250 can be integrated into a digital signal processor to implement data signal monitoring, signal processing, and signal compression encoding and decoding operations. Such a processor is chosen, for example, from the Texas Instruments TMS 320C5X family of digital signal processors.

The operation of the compression system of the present invention is now described. Still referring to Figure 2, when the voice call is established, the voice RELP codec 210 is initially assigned to the telephone signal. The CSP 200 monitors the telephone signal through the monitor signal 202 and the control unit 201 determines the type of voice band signal based on the detection of the answer signal from the modem. Each type of voiceband data has a particular identifiable modem answer signal. Table 1 summarizes some of the various typical modem origination and response characteristics that are well known in the art. Table 1 is for illustrative purposes and therefore is not intended to describe all possible modem features.

(Table goes to next page)

ES 2 231 899 T3

<td></td><td></td><td>space</td><td></td><td> 1320</td><td></td><td></td><td></td><td></td><td></td><td> 980</td><td></td><td></td><td> 2100</td><td> 1700</td><td></td><td></td><td></td><td></td><td></td><td></td><td></td>

<td rowspan="3">; ace of</td><td></td><td>expensive</td><td></td><td></td><td></td><td>or</td><td></td><td></td><td></td><td>or</td><td></td><td></td><td></td><td></td><td></td><td></td><td></td><td></td><td></td><td></td><td></td>

<td></td><td></td><td>skilled</td><td>or</td><td>or</td><td>or</td><td>or</td><td>or</td><td>or</td><td>OR</td><td> 100</td><td> 100</td><td> 100</td><td> 100</td><td> 100</td><td> 100</td><td> 100</td><td>or or</td><td> 100</td><td> 100</td><td> 100</td><td> 8</td>

ES 2 231 899 T3

<td>V. 32 4800</td><td>FAX Channel 300 BPS</td>

ES 2 231 899 T3

Returning to Figure 2, once the voiceband data type is determined, if high-speed data compression or ultra-high-speed data compression is required, the CSP 200 initiates a voice channel reassignment, and the method of dynamic time slot allocation used is described below. Control unit 201 sends a signal to channel forming processor 260 to form an RF communication channel with a predetermined number of time slots. In one embodiment of the present invention, a time slot is automatically assigned to the call, but this is not required. Channel shaping processor 260 examines memory to determine the number and RF carrier location of available RF time slots. If the channel forming processor locates the predetermined number of segments, the RF communication channel is formed from the predetermined number of RF time segments and the control unit 201 is notified of this. The control unit 201 then assigns a corresponding high-speed data codec or a corresponding ultra-high-speed data codec to the data signal, and the compressed data signal is assigned to the modulated multi-segment RF communication channel. about this.

If there are not enough time slots available, the control unit 201 is informed of this and an RF communication channel is formed from a single RF time slot, and the control unit 201 then assigns the CODEC of low speed data or the low speed FAX CODEC to the data signal. As noted above, an embodiment of the present invention automatically allocates a time slot when the telephone signal is received prior to forming a multi-time slot communication channel, and thus the telephone signal is already assigned to a segment in this. moment.

Dynamic allocation of time slots / bandwidths

Table 2 summarizes the time slot requirements for the types of signal compression:

TABLE 2

<td>Encoder</td><td># 4-phase segments</td><td># 8-phase segments</td><td># 16 segments phases</td>

<td>Low speed data</td><td> 2</td><td>N / A</td><td> 1</td>

<td>Low speed FAX</td><td> 2</td><td>N / A</td><td> 1</td>

<td>High speed data</td><td>N / A</td><td> 4</td><td> 3</td>

<td>Ultra-high-speed data</td><td>N / A</td><td>N / A</td><td> 4</td>

Since the high-speed encoder modulates data on both a 16-phase, three-segment channel and an 8-phase, four-segment channel, its compressed data desirably fits into one of the two channels with less bandwidth. Figure 3 shows the bit availability for the various types of channel of the embodiment of the described radio telecommunication system of Figure 1.

TABLE 3

<td>Mod level, segments</td><td>Mode</td><td>Start- what null</td><td>Pream- hoax</td><td>CW</td><td>Block TO</td><td>Block B</td><td>Final null</td><td>Bits / block what of data</td>

<td>16-PSK, 1</td><td>test voice / channel</td><td> 0</td><td> 5</td><td> 3</td><td> 80</td><td> 84</td><td> 8</td><td> 328</td>

<td>16-PSK, 3</td><td>voice (HSD)</td><td> 0</td><td> 5</td><td> 3</td><td> 262</td><td> 262</td><td> 8</td><td> 1048</td>

<td>16-PSK, 4</td><td>voice (UHSD)</td><td> 0</td><td> 5</td><td> 3</td><td> 352</td><td> 352</td><td> 8</td><td> 1408</td>

<td>8-PSK, 1</td><td>test channel</td><td> 0</td><td> 14</td><td> 4</td><td> 154</td><td> 0</td><td> 8</td><td> 462</td>

<td>8-PSK, 4</td><td>voice (HSD)</td><td> 0</td><td> 14</td><td> 4</td><td> 347</td><td> 347</td><td> 8</td><td> 1041</td>

<td>4-PSK, 2</td><td>test voice / channel</td><td> 0</td><td> 13</td><td> 6</td><td> 160</td><td> 173</td><td> 8</td><td> 328</td>

<td>BPSK, 1</td><td>Tuning</td><td> 0</td><td> 52</td><td>8 (UW)</td><td> 112</td><td> 0</td><td> 8</td><td> 112</td>

ES 2 231 899 T3

In Table 3, "null" indicates that no modulation is present, the preamble is a bit synchronization pattern, and "CW" means a code word that includes call control, call processing, and signaling information. Block A and Block B represent first and second 22.5 ms blocks of compressed voiceband data samples.

As seen in Table 3, the 8-phase, four-segment channel carries fewer bits than the 16-phase, three-segment channel. Thus, the compressed output block of the high-speed encoder of an embodiment of the present invention may occupy 1041 bits or less. Table 4A shows the bit allocation of the compressed output block of the high speed data encoder.

TABLE 4A

<td>Data</td><td>Bits per case</td><td>Quantity</td><td>Protected</td><td>Number of bits</td>

<td>Coded sample</td><td> 5</td><td> 180</td><td>Yes</td><td> 900</td>

<td>Coded gain</td><td> 6</td><td> 1</td><td>Yes</td><td> 6</td>

<td>Protected spare</td><td> 1</td><td> 6</td><td>YES</td><td> 6</td>

<td>Hamming parity</td><td> 7</td><td> 16</td><td>N / A</td><td> 112</td>

<td>Replacement</td><td> 1</td><td> 24</td><td>not</td><td> 24</td>

<td>Total per block</td><td></td><td></td><td></td><td> 1048</td>

In Table 4A the term "protected" indicates that a direct error correction (FEC) is applied to the bit stream. The ultra-high-speed encoder bit stream modulates a 16-phase, four-segment channel, of which 1408 bits are available for encoder data in each 22.5 ms time period.

Table 4B shows the bit allocation of the compressed output block of the ultra-high speed data encoder.

TABLE 4B

<td>Data</td><td>Bits per case</td><td>Quantity</td><td>Protected</td><td>Number of bits</td>

<td>Coded sample</td><td> 7</td><td> 180</td><td>Yes</td><td> 1260</td>

<td>Coded gain</td><td> 7</td><td> 1</td><td>Yes</td><td> 7</td>

<td>Protected spare</td><td> 13</td><td> 1</td><td>Yes</td><td> 13</td>

<td>Hamming parity</td><td> 7</td><td> 16</td><td>N / A</td><td> 112</td>

<td>Unprotected spare part</td><td> 16</td><td> 1</td><td>not</td><td> 16</td>

<td>Total per block</td><td></td><td></td><td></td><td> 1048</td>

The high-speed data and ultra-high-speed data compression techniques described below are embodiments of the present invention that may require multiple time slots for a communication channel, but other compression techniques can be developed in the same spirit as the one above. It is described here for other specific types of data signals that do not necessarily follow the voiceband modem characteristics described above. These other embodiments may also employ the dynamic time slot / bandwidth allocation method as used in the present invention.

The general method of dynamic time slot / bandwidth allocation is now described. Figure 3A illustrates the dynamic time slot / bandwidth allocation process as implemented in, for example, the CSP 200 of Figure 2. Referring to Figure 3A, when the call of voice, voice monitoring step 301 monitors the telephone for a data signal. In step 301, the RELP 210 codec is initially assigned to the telephone signal. However, when a sign of

ES 2 231 899 T3 data, decision step 302 determines the type of the voiceband signal based on the detection of the response signal from the modem.

If the data is low speed data or a low speed FAX, step 303 assigns the low speed allocation process to which, for example, a single RF carrier segment has been allocated. Step 304 then determines whether the data signal is a FAX or low speed data, and assigns the respective algorithm steps 305 and 306 of the low speed FAX codec 230 or the low speed data codec 220.

If the signal is of a high-speed data type in step 302, then the next step 307 requests a high-speed data channel from the channel formation process 260. In one embodiment of the present invention the channel formation process Channel 260 will require user / subscriber provisioning information to request the channel type. Another embodiment of the present invention can further determine from the modem signals whether the data signal requires the high-speed data compression method or the ultra-high-speed data compression method to request the correct channel type. .

Figure 3B shows the channel allocation process performed by channel shaping processor 260 upon request for a high speed data channel from step 307 of Figure 3A. The channel forming processor may be a base station radio processing unit (RPU) of the previously described prior art system example, and the RPU may allocate RF carrier time slots to subscriber communications to through a communication channel.

Starting at step 320 of Figure 3B, the processor typically allocates a voice channel for a telephone call; however, any initial process allocation can be chosen, such as described in US Patent No. 4,675,863. Next, step 321 checks for a request for a high-speed data channel from step 307 of Figure 3A. If no request is present, the assignment remains in the default mode, which is voice for this embodiment. If a request is present, step 322 checks the subscriber provisioning to determine if the subscriber is provisioned to accept a high speed data channel. If the subscriber is not provisioned to accept a high speed data channel, a low speed data / fax channel is assigned in step 323 using a predetermined number of segments.

If the subscriber is provisioned for a high-speed data channel, step 324 determines whether the subscriber is provisioned to accept an ultra-high-speed data channel of the ultra-high-speed type ("UHSD channel") (or if so required). If this is so, step 325 verifies if a predetermined number of RF carrier segments are available, and if so, step 326 then creates the UHSD channel. Step 325 may be embodied by a processor that checks a memory containing the current system channel assignments to find if a required number of sixteen-phase RF time slots (four for the exemplary embodiment) are available. If the required number of segments is not available, the process then tries to see if the channel can be created as a high speed data type ("HSD channel"), as described below in step 328.

If the subscriber's provisioning (or request) indicates that the high-speed data channel should not be formed as an ultra-high-speed type UHSD channel in step 324, step 327 checks whether the subscriber's request or provisioning indicates that the high-speed data channel should be formed as a high-speed type HSD channel. If not, the low-speed data channel is formed in step 323 as described above, but if the HSD channel is requested or provisioned, then step 328 checks if the predetermined number of time slots is available RF carrier for HSD channel.

Step 328 may be embodied by a processor that checks a memory containing the current system channel assignments to find if a required first number of time slots (sixteen phase RF time slots) is available (three for example implementation), and if not, whether a second required number of time slots (eight-phase RF slots) is available (four for the exemplary embodiment). If the required number of slots is available, the time slots are allocated and formed into HSD channel in step 329. If the high speed channel availability step cannot find the required number of channels, then step 323 simply assigns the low speed channel.

Returning to Figure 3A, in step 308 the process verifies the response to the high speed data channel request. If in step 308 the request is denied and no high-speed data channel has been formed, then steps 303 and the sequence to assign the low-speed algorithms are executed. If the request for the high-speed data channel is accepted, the high-speed channel availability step 309 determines what type of channel has been assigned. If the high-speed data channel corresponds to ultra-high-speed data, the coding algorithms of the ultra-high-speed data CODEC 250 are executed in step 310, and if the high-speed data channel corresponds to high-speed data, The high speed data CODEC encoding algorithms 240 are executed in step 311.

High-speed and ultra-high-speed CODECs

The high-speed codec 240 and the ultra-high-speed codec 250 provide compression of a bidirectional data channel of the present invention with sampled telephone signals (modulating telephone signals

ES 2 231 899 T3 by pulse code (PCM) in the exemplary embodiment) as the input signal and the output signal. The phone signals provided to the sample compression process are typically 64 kb / s A-law or Mu-law PCM, but 128 kb / s 16-bit integer samples, or other types, can be used using a conversion process. . The compression process compresses the 64 kb / s (or 128 kb / s) sample bit stream down to a lower data rate. The lower rate data is sent over the RF channel to the expansion process, which again expands the lower rate data to a reconstructed 64 kb / s (or 128 kb / s) sample bit stream. The goal of the encoder is for the synthesized or reconstructed samples to be an exact representation of the original sampled signal.

In PCM systems, analog voiceband signals are converted into a sequence of digital samples at a sampling rate of 8 kilosamples / second. The samples are 8 bits wide, resulting in 256 possible levels of quantization. When sampling analog signals, an important figure of merit is the signal to quantization noise ratio (SQNR). For a uniformly spaced quantizer, the SQNR is 6B - 1.24 dB, where B is the number of bits per quantized sample.

Therefore, an 8-bit uniform quantizer has an SQNR of 46.76 dB, which is excellent for speech signals. This SQNR is achieved only if the original analog signal has an amplitude that occupies the entire dynamic range of the quantizer. If the dynamic range of the original signal exceeds that of the quantizer, amplitude clipping occurs. This is a very undesirable type of distortion for voice modem and voice band signals. If the original signal has a smaller dynamic range than that of the quantizer, the resulting SQNR is less than the optimal 46.76 dB. For every dB that the dynamic range of the signal is less than the dynamic range of the quantizer, there is a 1 dB loss from the SQNR.

Since voiceband signals used in telephony have a wide dynamic range, a uniform quantizer may not be the optimal choice. Thus, non-uniform quantizers are used. There are two standards for non-uniform quantizers for PCM: Mu-law and A-law, and these standards are well known in the art and are described in Chapter 8, with Communication Systems, by Simon Haykin. Both techniques are logarithmically spaced quantizer levels in order to increase the dynamic range of the quantizers. Figure 4A shows the characteristics of the A-law quantizer.

The spacing between quantizer levels at high signal levels is greater than the spacing at low levels. The result is a more consistent SQNR on a sample-to-sample basis. Although the best SQNR for these quantizers is lower than the uniform 8- bit quantizer, these quantizers can provide good SQNR over a wider range of signal levels.

Figure 4B compares SQNR performance as a function of signal level for an A-law quantizer and a uniform 8-bit quantizer. Although the smooth quantizer shows superior performance at high signal levels, the A-law quantizer retains good SQNR over a wider dynamic range.

Voice band modems operate well in a telephone network employing Mu-law or A-law 64 kb / s PCM due to the wide dynamic range. The transmit output level of these modems is high to use the channels to their maximum capacity, but the telephone channels have variable losses in signal level. As a result, even when the modem output level is set high, the level elsewhere on the network can be significantly lower. The dynamic range of PCM compensates for this.

64 kb / s PCM compression at a lower data rate decreases the number of bits per sample and usually results in a significant decrease in SQNR. Distortion due to compression is minimized by the present invention by dynamically designing a quantizer to match the dynamic range of the input signal. Once the two dynamic ranges are paired, the samples are quantized using a quantizer with the newly defined level spacing.

Figure 4C illustrates a simple example of the compression method mapping signal samples from one quantization to another. A signal sample block 410 consists of three samples 411, 413, and 415. A first set of quantization levels 420 indicates the approximate value of sample amplitudes 412, 414, and 416. However, quantization levels require that a specified number of bits of information, five bits for the 20 displayed levels of the first quantization, be transmitted to a receiver in order to represent one of the levels of the first quantization. To send three sample values corresponding to the three samples 411, 413, and 415, fifteen bits are desirable.

The example of the method of the present invention defines a new set of levels for each block of signal samples based on the peak amplitude. As shown in Figure 4C, sample block 410 has a sample 413 exhibiting a peak amplitude value 414. The method defines a new set of quantization levels by defining peak amplitude 414 as the highest level value. and determines a predetermined number of level values below this amplitude. As shown in Figure 4C, this corresponds to five level values. For this new quantization, only three bits are needed to define a level value, but the peak amplitude value has to be sent also as a scale factor to indicate the relationship between the new quantizer level values and the level values. of original quantization. Consequently, five bits corresponding to the original peak amplitude value and nine bits (three per sample) are transmitted for the sample block 410,

ES 2 231 899 T3 that is, fourteen bits are required. The example shows that one bit less is sent; however, if there are ten samples in the block, the original quantization method requires fifty bits to be sent, but the new quantizer only requires thirty-five bits to be sent.

The following describes embodiments designed for the Mu-law and A-law standards. However, the techniques discussed are easily extended to any system for receiving quantized samples with a non-uniform compression-expansion quantizer.

High speed data CODEC

Figure 5A is a high-level block diagram of the high-speed data encoder. The exemplary encoder transforms data between a 64 kb / s PCM and a 46.58 kb / s direct error correction (FEC) encoded compressed data stream. The compressed data rate is 40.267 kb / s and the remaining transmitted bit stream is used for error correction.

As shown in Figure 5A, the high speed data encoder of the present invention includes an optional buffer 510, a PCM expander 520, a gain calculation process 522, a delay 521, a data sample quantizer 523 and an optional transmit encoding process 530. The transmit encoding process 530 further includes a FEC encoder 532 and an interleaver 531.

Optional buffer 510 contains a predetermined number of samples to create a sample block for high speed data compression processing. Alternatively, the samples can be received in a block format. The PCM Expander 510 converts A-law or Mu-law PCM samples to linear samples. The gain calculation process 522 calculates the quantized gain value for the sample block, and the data sample quantizer uses the quantized gain value to create a uniformly spaced quantizer with quantization level values scaled by the value. quantized gain. The delay shows that the quantized gain value is determined before the compression process creates encoded quantized samples, and the transmission encoding process 530 is used to provide error correction encoding for the transmission of the encoded quantized gain and the encoded quantized samples.

The operation of the high speed data compression encoder is now described. As shown in Figure 5A, the 64 kb / s PCM samples (A-law or Mu-law) are received by a buffer 510. The buffer 510 provides the PCM samples as 22.5 millisecond sample blocks. . At the PCM rate of 8 kilosamples / second, each block contains 180 samples. The received PCM frame is fed to PCM expander 520, which converts the Mu-law or A-law samples to 16-bit linear samples (16-bit integer samples).

The resulting block of linear samples, which are 16-bit integer samples in the exemplary embodiment, is fed to the gain calculation process 522, which finds the sample in the block with the highest amplitude value (absolute value). The amplitude of this sample determines the quantized gain value for the block. The quantized gain value can be the amplitude value, the difference between the maximum value in the sample and the largest amplitude in the block, or a multiplier value. The quantized gain value is quantized using a 64-level logarithmically spaced quantizer. The gain calculation process 522 provides both the quantized gain and the encoded quantized gain value. The encoded quantized gain value is a 6-bit number representing one of the 64 levels in the logarithmically spaced gain quantizer.

The quantized gain value from the gain calculation 522 and the sample block from the PCM expansion process are supplied to the data sample quantizer 523. The delay 521 is displayed to indicate that the gain calculation process 522 is to complete the task on the block before the samples are compressed by the 523 data sample quantizer. The data sample quantizer 523 quantizes the 180 samples in the block using a 32-level uniformly spaced quantizer. Quantizer levels are dynamically adjusted on a block-by-block basis using the quantized gain value. Thus, the levels of the uniformly spaced quantizer range from + quantized gain value to -quantized gain value for the current set of 180 samples. The sample quantizer outputs only the 5-bit encoded representation of the 180 samples, since compression does not require the actual quantized values.

The encoded quantized gain and encoded quantized samples are optionally fed to the transmission encoding process 530, which includes the interleaver 531 and the FEC encoder 532. The FEC encoder 532 is an extended Hamming encoder (64,57) and the Hamming code is capable of correcting a single bit error and detecting a double bit error in each 64-bit block. FEC encoder 532 receives the encoded quantized gain and encoded quantized samples and supplies them to interleaver 531, and interleaver 531 outputs encoded compressed data. The interleaver of an exemplary embodiment of the present invention is a 16 * 64 bit block interleaver.

Figure 5B shows an exemplary embodiment of transmission encoding process 530 that includes interleaver 531 and FEC Hamming encoder 502. A 64-by-16-bit block is shown. Each of the 16 rows represents a single 64-bit extended Hamming code word. Data is entered into the encoder by

ES 2 231 899 T3 reading in the interleaver block from left to right through the rows starting with bit 0 of code word 0 and ending with bit 63 of code word 15. Bit positions are skipped (column) 0, 1,2,4, 8, 16 and 32 and are filled with zero. After filling the interleaver 531, a Hamming encoding is performed by the FEC encoder 532 on the 57 data bits of each row. Hamming parity bits are inserted at bit positions 1,2, 3, 8,16, and 32, as shown in the diagram. The parity check bit is inserted into bit position 0. The parity bits and parity check bits for all 16 codes can be computed at the same time using a 10-bit wide exclusive 0 function (XOR function). The Pi parity bits are computed as follows:

Pi = XOR bit codeword [k] i = 0..6 (k-1) & 2 '+ 0, where is a bitwise binary Y function.

After the parity bits are inserted into their bit positions, the PC parity check bits (one bit for each code) are computed as follows:

PC = XOR codeword bit [k] k = 1

After the parity bits have been computed and inserted, data is read from the interleaver from top to bottom of the columns starting at code word 0, bit 0 and ending with code word 15, bit 63.

Figure 6A is a high-level block diagram of the high-speed data decoder in accordance with an exemplary embodiment of the present invention. The high speed data decoder implements the inverse of the data compression process of the high speed data encoder and the decoder includes an optional transmit decoding process 601, a frame gain decoder 610, a data sample decoder 620, a PCM compressor-expander 630, and a buffer 640. The transmit decoding process 601 includes a de-interleaver 603 and a FEC decoder 602.

The operation of the high speed data decoder is now described with reference to Figure 6A. The received compressed data is optionally fed to deinterleaver 603, which is a 16 * 64 bit block deinterleaving process. The output of deinterleaver 603 is fed to FEC decoder 602, which is an extended Hamming decoder (64,57). The Hamming decoder can correct a bit error and detect 2 bit errors per block. Figure 6B shows the deinterleaver and Hamming decoding process of one embodiment of the present invention. Data is read input into deintegrator 603 from top to bottom starting with bit 1 of code word 0 and ending with bit 63 of code word 15. The syndrome is computed as follows:

Compute parity bits:

Pi = XOR bit codeword [k] i = 0..5 (k-1) & 2 '+ 0, where is a bitwise binary Y function

Syndrome = concatenation P5 I P4I P3I P2IP1IP0

The parity check bits (one bit for each code) are computed as follows:

PC = XOR codeword bit [k] k = 1

The numerical representation of the syndrome indicates the bit position (if any) where a bit error has occurred. When a bit error has occurred, the bit is inverted (corrected) if the parity check bit is set for that code. Otherwise, it is assumed that there are 2 (or more) bit errors in the code and that the syndrome is incorrect. If the syndrome is zero, no bit error has occurred. As in the case of the encoder, the parity bits and parity check bits for all 16 codewords can be computed at the same time using a 16-bit wide exclusive-OR operation.

Returning to Figure 6A, the decoded data from the FEC decoder 602 consists of the encoded quantized samples and the encoded quantized gain. The encoded quantized gain is supplied to the gain decoder 610, which reads the quantized gain value from a table using the gain

ES 2 231 899 T3 quantized encoded as the table entry index. As mentioned above, the encoded quantized gain represents a level value of a logarithmically spaced 64-level quantizer.

The quantized gain value is supplied to the data sample dequantizer 620, where it is used to scale the level values of a level table of a 32-level uniform quantizer. The scaled quantizer table decodes the encoded quantized samples as a block of linear quantized samples.

The linear quantized sample block is converted to a PCM sample block (A-law or Mu-law) by the PCM compression-expansion process 630. The PCM sample block is then optionally supplied to buffer 640, which supplies the PCM samples as a 64 kb / s output signal.

The ultra-high speed CODEC

Figure 7A is a high-level block diagram of the ultra-high-speed data encoder. The ultra-high-speed data encoder performs data compression and expansion of the ultra-high-speed voiceband modem signals. The encoder transforms data between a 64 kb / s PCM and a 62.58 kb / s FEC encoded compressed data stream. The actual compressed data rate is 56.311 kb / s and the remaining bit stream is used for error correction data. The ultra-high-speed codec is similar to the high-speed codec.

As shown in Figure 7A, the ultra-high speed data encoder of the present invention includes an optional buffer 710, an optional sample format preprocessor 720, a gain calculation process 722, a delay 721, a quantizer of data samples 723 and an optional transmit encoding process 730. The transmit encoding process 730 further includes an FEC encoder 732 and an interleaver 731.

Optional buffer 710 contains a predetermined number of samples to create a sample block for ultra-high speed data compression processing. Sample format preprocessor 710 removes A-law or other standard transmission formatting from PCM samples and also converts the sample values to a predetermined numeric format, such as their decimal equivalents, for convenience in subsequent processing. The gain calculation process 722 calculates the quantized gain value for the sample block, and the data sample quantizer uses the quantized gain value to create a set of quantizer levels with predetermined spacing and with quantization level values set. scaled by the quantized gain value. The delay shows that the quantized gain value is determined before the compression process creates encoded quantized samples, and the transmission encoding process 730 is used to provide error correction encoding for the transmission of the encoded quantized gain and the transmissions. encoded quantized samples.

The operation of the ultra-high speed data compression process is now described. The 64 kb / s PCM samples (A-law or Mu-law) are fed to buffer 710. Buffer 710 supplies the PCM samples as 22.5 millisecond sample blocks. At the PCM rate of 8 kilosamples / second, each block contains 180 samples.

Unlike the high speed codec, the ultra high speed codec does not convert PCM samples to linear samples. In contrast, the 8-bit PCM data is converted into a predetermined type of format for the representation of the samples. In the exemplary embodiment, no operation for format conversion is required for Mu-law but for A-law the sample format preprocessor 720 converts the samples to a predetermined level value format before subsequent quantizer processing. As is apparent to a person skilled in the art, the Mu-law samples could be converted to an A-law representation or, in another embodiment, both formats could be converted to a third predetermined format.

In the ultra-high speed codec it is desirable that the type of PCM compression is the same at both the transmitting and receiving ends of the link. Otherwise, without additional processing, the differences between the Mu-law and A-law characteristics can cause a non-linearity in the end-to-end characteristics of the compression encoding.

The block of samples received in the predetermined sample format is fed to the gain calculation process 722, which finds the sample in the block with the highest amplitude value (absolute value). The amplitude of this sample determines the quantized gain for the block. Quantized gain requires 7 bits, since the amplitude sign bit is not used.

Table 5 shows how numbers are represented in A-law and Mu-law norms. The absolute value of the sample corresponding to these respective representations is determined and the maximum amplitude is calculated.

ES 2 231 899 T3

TABLE 5

<td>Give yourself number</td><td>Law A Equiv</td><td>Law A Hex</td><td>Μ Equiv law</td><td>Μ Hex law</td>

<td> 1</td><td> 129</td><td> 81</td><td> 254</td><td>FAITH</td>

The quantized gain from the gain computing process 722 and the 2's complement block are supplied to the data sample quantizer 723 after the quantized gain value is calculated, as shown by the presence of delay 721.

The data sample quantizer 723 creates a new quantizer with a set of quantizer levels from the A-law or Mu-law sample block. The following discussion describes how the new quantizer is determined for a sample block. . The A-law quantizer divides the input amplitude range into 7 segments and the Mu-law quantizer divides the input amplitude range into 8 segments. For convenience, the following discussion describes the A-law process with 7 segments, but it is obvious to one skilled in the art to extend the discussion of A-law to compression of Mu-law samples.

Each segment (except the first) has a range of amplitudes that is half the range of the next, and each segment (except the first) has 16 quantization level values. As a result, the size of the quantizer step in each segment is twice that of the previous one. Table 6 lists the A-law quantizer segments along with their amplitude ranges and step sizes from an exemplary embodiment.

TABLE 6

<td>Segment number</td><td>Input amplitude range</td><td>Normalized amplitude range</td><td>Standard step size</td><td>Law Code A</td>

The samples representing the input data signal can span the entire dynamic range of the A-law quantizer and the A-law quantizer is converted into a new quantizer by removing selected levels from among the levels of the A-law quantizer. The following illustrates the process if the resulting new quantizer has evenly spaced level values and all segments are used to represent a block of samples. The step size of the last segment, 1/32, is the largest step size in the quantizer and therefore

ES 2 231 899 T3 Therefore, all quantizer level values are held in the last segment. The sixth segment has a quantizer level value step size of 1/64. A step size of 1/32 in the seventh segment determines that every second quantizer level in the sixth segment is removed, resulting in a step size of 1/32. Similarly, this process is repeated for the fifth through third segments. The second and first segments combined span only a 1/32 range, and therefore none of the quantizer levels are retained. This results in 31 positive levels and 31 negative levels, and a zero level is retained to separate the first positive segment and the first negative segment, giving a uniform 63-level quantizer.

The process then computes the peak amplitude of the sample block and determines which A-law segment contains that amplitude. For that data block all segments above this "peak segment" are ignored. The step size of the peak segment defines the step size of the uniform quantizer. Thus, in the resulting smooth quantizer for the block, all levels of the quantizer are held at the peak level, half of the levels are held in the immediate lower segment, and quantizer level values are assigned until the last one is reached. segment or no more quantizer level values are available.

Figure 9 shows the method of operation of the ultra-high-speed quantizer, a 128-level quantizer, of an embodiment of the present invention.

At step 904, the method receives a block of compressed-expanded samples (such as an A-law or a Mu-law compression-expansion).

In step 906, the peak amplitude sample in the block and the corresponding segment are determined, and the peak amplitude value is the peak segment.

In step 910, each peak segment quantizer level value is held.

In step 912, unless level zero has been reached, all 16 levels of the next segment are held.

In step 914, unless level zero is reached, all 16 levels are retained in the next segment.

In step 916, unless level zero is reached, every second level value (8 level values) is retained in the next segment.

At step 918, unless level zero is reached, four levels are held in the next lower segment.

In step 920, unless level zero is reached, 2 levels of the next lower segment are held.

In step 922, unless the zero level is found, 1 level of the next lower segment is held.

At step 924 the zero level is held.

Finally, in step 926, the negative levels are created using equal magnitudes as positive values, but of opposite sign, setting a sign value.

The peak amplitude (7 bits) and the 180 7-bit encoded samples constitute the compressed output of the ultra-high-speed encoder compression process.

Returning to Figure 7A, the encoded quantized gain and encoded quantized samples are supplied to the transmit encoding process 730. The exemplary embodiment of the transmit encoding process 730 includes the FEC capacitor 732, which is, for example, a Hamming encoder (87.80). The Hamming code is capable of correcting a single bit error in the 87-bit block. The FEC encoder supplies the compressed direct error correction encoded uniformly quantized data samples to interleaver 731, which is, for example, a 16 * 87 bit block interleaver. Interleaver 731 supplies coded compressed data for modulation on the RF communication channel.

FIG. 7B is a block diagram of the transmission coding process of the exemplary embodiment of the ultra-high-speed data encoder. An 87-by-16-bit block is displayed. Each of the 16 bit rows represents a single 87-bit Hamming code word. Data is entered into the encoder by reading into the interleaver block from left to right through the rows starting with bit 1 of code word 0 and ending with bit 86 of code word 15. Positions are skipped of bits (columns) 1, 2, 4, 8, 16, 32 and 64 and these are filled with zeros. The last column / word of the interleaver block receives special treatment. It only contains data in its first three rows / bit positions. The remaining rows / bit positions are filled with zeros.

After filling the interleaver, a Hamming encoding is performed on the 80 data bits in each row. Hamming parity bits are inserted at bit positions 1, 2, 4, 8, 16, 32, and 64, as shown in the diagram. Parity bits for 6 codes can be computed at the same time using an exclusive OR function

ES 2 231 899 T3 (XOR function) 16 bit width of the DSP. The Pi parity bits are computed as follows and as shown in Table 7:

Pi = XOR bit codeword [k] i = 0..6 (k-1) & 21 + 0, where "&" is a bitwise binary AND function TABLE 7

<td>Parity bit</td><td>XOR setting</td>

After the parity bits have been computed and inserted, data is read from the interleaver from top to bottom of the column starting at codeword 0, bit 1 and ending with codeword 15, bit 87.

Table 8 shows the interleaver block. There are 88 words numbered from 0 to 87. The first word is not used, but is kept by similarity to HSD. The first word is not transmitted. The numbers 0 to 1266 represent the 1267 bits of the 181 words. "P" in Table 8 means parity.

(Table goes to next page)

ES 2 231 899 T3

TABLE 8

Figure 8A is a block diagram of the ultra-high speed data decoder of the present invention. The data expansion process is the reverse of the data compression process and the decoder includes an optional transmit decoding process 801, a gain decoder 810, a decoder of65 data samples 820, an optional sample format reprocessor 830, and an optional buffer memory 840. The optional transmit decoding process 801 includes a deinterleaver 803 and a FEC decoder 802.

ES 2 231 899 T3

As shown in Figure 8A, the received encoded compressed data is supplied to transmission decoding process 801 to remove transmission encoding and correct transmission errors. The transmit decoding process 801 of the exemplary embodiment of the present invention includes the deinterleaver 803, which is a 16 * 87 bit block deinterleaver. The output of the deinterleaver 803 is supplied to the FEC decoder 802, which is a Hamming decoder (87,80). The Hamming decoder can correct a bit-per-block error.

FIG. 8B shows an embodiment of the ultra-high-speed data decoder transmission decoding process of an embodiment of the present invention, including de-interleaving and Hamming decoding. The encoded compressed data is read input into the deinterleaver from top to bottom starting with bit 1 of code word 0 and ending with bit 86 of code word 15. Special handling is required for the last column / word.

The numerical representation of the syndrome indicates the bit position (if any) where a bit error has occurred. When a bit error has occurred, the bit is inverted (corrected). If the syndrome is zero, no bit error has occurred. As in the ultra-high-speed data encoder, the parity bits for up to 16 code words can be computed at the same time using a 16-bit wide exclusive-OR operation (XOR operation).

The syndrome is computed as follows:

Compute parity bits:

Pi = XOR bit codeword [k] i = 0..6 (k-1) & 2 '+ 0, where is a bitwise binary Y function

Syndrome = concatenation P6I P5I P4I P3 I P2I P1 I P0

The decoded data from the FEC decoder 801 consists of encoded quantized samples and encoded quantized gain. The encoded gain is fed to the gain decoder, which supplies the quantized gain value to the data sample dequantizer 820.

The data sample quantizer generates a lookup table containing the A-law (or Mu-law) quantizer levels corresponding to the 7-bit coded samples using the quantized gain value (the peak amplitude sample of the block) . The quantizer is created using exactly the same procedure that has been described in the section on the ultra-high-speed data encoder, in which the lookup table has 256 entries, each of the entries corresponding to one of the 128 possible quantized samples encoded. However, the lookup table is used in the opposite way. Once the lookup table with 128 entries of the possible values of coded quantized samples is generated, the corresponding PCM samples are found in the table by indexing the corresponding coded quantized samples (7-bit codes) with respect to the input of the table.

As shown in Figure 8A, if A-law compression-expansion is desired, an optional sample format reprocessor 830 transforms the decoded sample block into a desired sample format, such as A-law. For A-law or Mu-law, the block of decoded samples corresponding to the reconstructed ultra-high-speed data samples is supplied to the output buffer 840, which provides a 64 kb / s PCM compressed-expanded signal as exit sign.

Although preferred embodiments of the invention have been shown and described herein, it will be understood that such embodiments have been provided by way of example only. Numerous variations, changes, and substitutions will occur to those skilled in the art.

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19960743749	United States of America	–	1996-11-07
74374996	United States of America	A	1996-11-07

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CA2242346A1	Canada	A1	1998-05-14
CA2405527A1	Canada	A1	1998-05-14
CA2476714A1	Canada	A1	1998-05-14
WO9820696A2	World Intellectual Property Organization (WIPO)	A2	1998-05-14
AU5245398A	Australia	A	1998-05-29
EP0880868A2	European Patent Office (EPO)	A2	1998-12-02
WO9820696A3	World Intellectual Property Organization (WIPO)	A3	1999-02-18
CN1217860A	China	A	1999-05-26
JP2000504524A	Japan	A	2000-04-11
HK1020826A1	Hong Kong, China	A1	2000-05-19
US6111870A	United States of America	A	2000-08-29
US6385189B1	United States of America	B1	2002-05-07
US2002131391A1	United States of America	A1	2002-09-19
US2002131394A1	United States of America	A1	2002-09-19
US2002136194A1	United States of America	A1	2002-09-26
US2002136195A1	United States of America	A1	2002-09-26
US2002163888A1	United States of America	A1	2002-11-07
CA2242346C	Canada	C	2002-12-31
US6526383B1	United States of America	B1	2003-02-25
CN1420706A	China	A	2003-05-28
CN1420707A	China	A	2003-05-28
US6574207B2	United States of America	B2	2003-06-03
CN1110982C	China	C	2003-06-04
CN1422100A	China	A	2003-06-04
CN1423425A	China	A	2003-06-11
US2003144836A1	United States of America	A1	2003-07-31
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JP2004236343A	Japan	A	2004-08-19
US6792403B2	United States of America	B2	2004-09-14
EP1458104A2	European Patent Office (EPO)	A2	2004-09-15
EP0880868B1	European Patent Office (EPO)	B1	2004-11-03
AT281745T	Austria	T	2004-11-15
ATE281745T1	Austria	T1	2004-11-15
DE69731459D1	Germany	D1	2004-12-09
DK0880868T3	Denmark	T3	2005-03-14
PT880868E	Portugal	E	2005-03-31
CN1197429C	China	C	2005-04-13
US6888815B2	United States of America	B2	2005-05-03
ES2231899T3This record	Spain	T3	2005-05-16
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US7035291B2	United States of America	B2	2006-04-25
CN1770796A	China	A	2006-05-10
EP1458104A3	European Patent Office (EPO)	A3	2006-05-17
US7061885B2	United States of America	B2	2006-06-13
CA2476714C	Canada	C	2006-07-25
US7126934B2	United States of America	B2	2006-10-24
CN1901567A	China	A	2007-01-24
US2007036124A1	United States of America	A1	2007-02-15
HK1100114A1	Hong Kong, China	A1	2007-09-07
JP4024767B2	Japan	B2	2007-12-19
JP4046766B2	Japan	B2	2008-02-13
CN100456779C	China	C	2009-01-28
EP2276211A1	European Patent Office (EPO)	A1	2011-01-19
EP1458104B1	European Patent Office (EPO)	B1	2011-06-15
AT513437T	Austria	T	2011-07-15
ATE513437T1	Austria	T1	2011-07-15
ES2368207T3	Spain	T3	2011-11-15
EP1458104B9	European Patent Office (EPO)	B9	2012-01-25
US8503372B2	United States of America	B2	2013-08-06
US2013315225A1	United States of America	A1	2013-11-28
EP2276211B1	European Patent Office (EPO)	B1	2013-12-25
DK2276211T3	Denmark	T3	2014-03-03
ES2453901T3	Spain	T3	2014-04-08
US9295057B2	United States of America	B2	2016-03-22

Method and apparatus for compressing and transmitting high speed data

Abstract

Term

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DESCRIPTION

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Priority claims2

Members71

Numbers

Titles2

Classification