JP2003509949A - Circuit used for communication system and receiver unit - Google Patents

Abstract

The present invention relates to a code division multiple access (CDMA) communication system, and more particularly to a demodulator (11) of a receiver (10) by means of reducing the dynamic range of a long section of a received signal. ) Simplifies and improves the signal processing in receiver (10). This can reduce the number of bits required to encode the signal. This is achieved by signal processing means (16) located in the signal input path before the demodulator (11). The signal processing means (16) includes an improved digital AGC unit (161) for scaling the samples of the received signal to a common and constant power level, and a reduced number of bits for the scaled signal samples. And a quantizer (162) for converting into a sample. These reduce the transmission bandwidth of the receiver unit and reduce the complexity associated with signal processing in the demodulator.

Description

Detailed Description of the Invention

FIELD OF THE INVENTION The present invention relates generally to communication systems, and more particularly to a receiver unit including means for automatic gain control of received signals, preferably code division multiple access (CD).
MA) receiver unit used in a communication system.

BACKGROUND OF THE INVENTION In a code division multiple access (CDMA) based communication system, a plurality of base stations cover a given geographical area, and the base stations are users located in such areas. To or from a user. A feature of the CDMA system is that a common communication medium is shared by different users by assigning a unique and unique code sequence such as a pseudo noise code to the uplink channel and the downlink channel between the base station and the user equipment. Is to be done.
These code sequences are used by the transmitter to convert the signal into a wideband spread spectrum signal. For example, in a receiver unit, such as a base station or user equipment, a demodulator reconverts the aforementioned wideband signal transmitted from a particular transmitter back to the original band. When re-converting, the same code sequence used in the transmitter is used. On the other hand, the signals encoded by the different codes remain wideband signals. Therefore, signals encoded with different codes will be interpreted by the receiver as part of the external background noise.

However, for a physical wireless communication channel, the signal is thus subject to attenuation due to distance and propagation effects such as multipath attenuation and shadowing due to obstacles on the wireless path. Therefore, the communication medium needs to be modeled as a multipath channel with different delay times and attenuation indices due to different propagation paths.

Furthermore, another aspect unique to the multiple access technology used in CDMA is potentially associated with the fact that all users transmit wideband signals simultaneously and using the same band. This fact and the above-mentioned propagation effects result in the base station receiving wideband signals from various users within a large dynamic range of up to 80 dB depending on the received signal power. Further, the signal power transmitted and received from a single user may be altered during transmission, and the transmission from individual users may be magnified as the power may change due to the receiving antenna. It will follow the dynamic range. Therefore, in order to receive signals from all users present in the coverage area, such as a base station, the base station must not only calculate the total uplink transmit power received per user, but also all users. It is necessary to provide a means to monitor and control the total uplink transmit power received from the. These are monitored and controlled so as to reach the base station with virtually the same average power without depending on the propagation effect of the signal. The total interference level is also monitored and controlled so that it does not become so high as to reduce system capacity. These are achieved by estimating the total received power at each base station antenna and using the estimate to adjust the output power of the transmitting user equipment directed to each base station.

In other respects, application to an automatic gain controller (AGC) unit in the analog part of the receiver allows processing of signals with a large dynamic range. According to Japanese Patent Laid-Open No. 8-335928, for example, a receiver of a CDMA mobile communication system is shown, that is, a gain is controlled according to the maximum amplitude of an analog baseband signal, and a constant output signal (S30). ) Output AGC
A unit (30) is included.

SUMMARY OF THE INVENTION A first object of the present invention is preferably a CDMA based communication system, with a circuit capable of simplifying and improving the signal processing of the demodulator part of a receiver unit. To achieve the method.

Another object of the invention is to sequence digital signal samples by means of reducing the number of bits used for coding the samples without losing any information carried by the signal. To simplify and improve the process.

Another object of the invention is also to ensure that a reliable signal is used for demodulation, in principle receiving from several signal inputs, eg a base station antenna. To allow the same signal to be used by the demodulator.

Another object of the present invention is further to achieve an improved Digital Automatic Gain Controller (AGC) unit that is less sensitive to external noise interference and requires minimal hardware.

These and other objects of the invention are achieved by signal processing means comprising at least a digital AGC unit followed by a quantizing unit, which is arranged before the demodulator of at least one of the signal input paths. . The digital AGC unit scales the sequence of signal samples to a common and constant power level. The quantization unit can then transform the scaled signal samples into samples that use the reduced number of bits. In order to use AGC efficiently, the control error is determined and the determined control error is used to maintain the signal quality at the demodulator input.

In the present invention, signals input from various antennas may have a long-range dynamic range, that is, a large difference in signal power may occur due to variations in reception characteristics. It usually has a much smaller short range dynamic range. This is especially true for CDMA based communication systems. This implies that the corresponding digital input signal must be coded as a sample with many redundant bits in order to be able to represent a large dynamic range. The number of coded bits can be reduced by converting the input signal to a common power level which represents only the short range dynamic range.

Accordingly, a first advantage of the present invention is that it includes a method and apparatus that reduces the number of bits required to represent the signal samples transferred to the demodulator.

Therefore, another advantage of the present invention is that it reduces the transmission bandwidth required at the transmitter unit, ie the number of transmission bits per second.

Therefore, yet another advantage of the present invention is that the processing of the signal samples in the subsequent demodulator is considerably simplified, which reduces power consumption and chip real estate. Implicitly.

Furthermore, another advantage of the present invention is that it can provide an improved digital AGC unit that is less sensitive to external interference noise. C
In a DMA-based communication system, this AGC unit can also be implemented by reusing existing power estimation means for the received signal.

Furthermore, another advantage of the present invention is that the demodulator can switch in a fast manner to the antenna that receives the best signal instantaneously, eg in the case of softer handover. Is effective for.

Other objects, advantages and novel features of the present invention will become apparent by considering the following detailed description in connection with the accompanying drawings and claims.

DETAILED DESCRIPTION FIG. 1 generally illustrates a base station, eg, a base station in a CDMA-based communication system, and specifically a base station receiver 10 that transfers a received signal to a demodulator 11 thereof.
Is shown. A base station is usually divided into many sectors and equipped with at least one antenna per sector, but preferably two antennas in order to improve the reception quality of the signal using the diversity effect. Equipped with,
Providing communication services to user equipment in the cell.

Transmissions from user equipment located in one of the sectors are mainly received by antennas directed to a particular sector, but also by antennas of adjacent sectors. Thus, the base station can use the various received copies of the signal transmitted by the user equipment. For example, an important pre-requisite for softer handovers is that when a better quality received signal is detected from another sector antenna, the base station switches to that other sector antenna. The signal quality of each received signal is quite different and is received at different power levels. This is because the actual radio channel is affected by various propagation characteristics and attenuation characteristics. Therefore, a large dynamic range covers various power levels. Furthermore, the difference may be due to the fact that signals from outside the proper range of the antenna are received. Therefore, the present invention extracts an optimized signal that can be used to demodulate the original user information from among the various received signal copies for demodulation of the base station receiver 10, and in particular of the receiver 10. It is intended to be used in part 11 of the vessel.

Therefore, the base station receiver section 10 has a separate input path for the uplink signal transferred to the demodulator 11 for each antenna. First, the received high frequency signal is pre-processed by the amplifying and filtering means 12 and the intermediate frequency (IF) 13
Is down-converted into a 16-bit signal sample sequence by the analog-digital converter 14 and the low-pass filter 15 following it.
According to the present invention, the digitized input signal is converted to a sufficiently high target power level by using the signal processing means 16. Since this target power level is common to each signal input path 17, it can be applied to the demodulator 11 as described above. For each signal input path, the signal processing means 16 comprises a digital AGC unit 161 for scaling the input sequence and a quantizing unit 162 for reducing the number of bits needed to code the signal samples. Have at least. Optionally, the adaptation unit is a conventional analog AGC
Unit 163 may be included and may be applied, for example, as a preceding signal attenuation unit for analog signals.

FIG. 2 shows details of different parts of the signal processing means 16 in the present invention and how they cooperate in accordance with a preferred embodiment of the present invention. The signal processing means can be divided into a large number of blocks. The main block is the scaling block 21, which scales the input signal based on the power estimate of the signal from the power estimation unit 22. The quantization block 23 transfers the scaled output signal. Furthermore, the signal adaptation unit comprises an offset compensation means 24 and optionally a conventional analog AGC unit 25.

A scaling block 21 is provided at the center of the signal processing means. The scaling block 21 determines the scaling factor from the estimated power value for the received input signal X [n] and the input to the output signal Y [n] carrying the same information to the latter stage. Scaling means 21 for scaling the signal X [n]
Includes 2 and. For the input signal, which is the sequence of digital signals X _k [n], where k denotes the index of the time interval, the scaling is such that each of these signal samples has an appropriately determined scaling factor α _This means that multiplication by _k adjusts to a common target power level σ ² _ref , that is, Y _k [n] = α _k · X _k [n]. Therefore, the scaling block 21 compensates for variations in the average power level of the samples of the input signal, while
It reduces the dynamic range of a long section of a signal that does not include any user information and simply represents the strength of the wireless channel. This will produce a scaled output signal Y [n] with a total dynamic range corresponding to the short term dynamic range of the input signal X [n]. The short range dynamic range is generally much smaller than the long range dynamic range.

Therefore, in the coding of the signal samples described above, it is no longer necessary to express the redundancy caused by the significantly different signal power levels. Alternatively, the quantization unit 23 may reduce the number of coded bits per sample to a number sufficient to represent signal changes within the short term dynamic range.

Assuming that the sequence of input signal samples X _k [n] has an approximately constant variance σ ² _k , multiply by a scaling factor that is a constant value over the k th time interval. Scaling is performed by doing. Here, the scaling factor is defined as follows.

Α _k = √ (σ ² _ref / σ ² _k ) To calculate the scaling factor α _k for the input signal X _k [n], all time intervals during the calculation of the variance σ ² _k It is necessary to store the input data stream in the buffer. However, if the probability of significant fluctuations in the average received signal level, ie the variance between two consecutive frames, can be assumed to be small, then by using the estimate of variance obtained from the previous frame, Y _{Since k} [n] can be calculated as k, it is not necessary to store it in the buffer. Therefore, in the method according to the preferred embodiment of the present invention, the following equation can be used to derive the scaled output signal Y _k [n].

Y _k [n] = α _k−1 · X _k [n] In the above equation, the scaling function is expressed in the form of multiplication. AGC
In Scaling, the sample value is used as the basis for calculation. Therefore, difficult requirements may arise in the multipliers that perform the actual scaling. However, if the scaling factor could be quantized to a value expressible as a "power" of 2, these requirements would be greatly relaxed. In this case, the scaling factor α _k can be replaced by the shift factor β _k .

[0027] ^{_{α k = 2 βk <=> β k}} = log 2 α k scaling can be performed by left shifting the sample values of the signal in accordance with a shift factor beta _k. To avoid the large number of steps associated with the scaling factor α _k associated with quantization, the above factors may be quantized to a value that can be represented as a “power” of 2. In this case, the scaling factor α _k is expressed as a set of shift values {β _k , i}, which consist of exponents of the sum terms. Scaling is performed by left shifting the signal samples according to the position number indicated by each of the shift factors in the set, and then adding the shifted parts.

In the scaling block as described above, it is assumed that the average value of the input signal X [n] is 0. This assumption is necessary because the scaling operation is performed by multiplication. However, in reality, the hardware of the radio receiver may introduce a bias into the input signal that can be regarded as a time-varying process with a very large time constant. This deviation is eliminated by introducing the offset compensation block 24. This block includes estimation means 241 for estimating the average value of the signal. The estimation means uses a 1-tap low-pass filter having a large time constant. By subtracting the estimated mean value from the input signal,
The signal offset is removed.

An analog AGC unit 25 may optionally be introduced to reduce the dynamic range of the overall received signal. The unit is designed as an attenuator controlled by the power estimate from the power estimation unit 22. Control device 2
51 performs some level of attenuation depending on the estimated power level of the signal. The analog AGC unit should at least be designed as a step attenuator that includes an operating state that performs some damping and a non-operating state that performs small damping. The control device 251 activates the attenuating operation if the power level of the signal exceeds a certain threshold level, and sets the inactive state if the signal level drops below the threshold value. Activate. For example, the attenuator attenuates signal signals that require a very large dynamic range.
It is possible to avoid occurrence of undesired clipping in the analog-digital converter in the subsequent stage.

As explained above, the scaling factor for each signal sample is defined using an estimate of the variance σ ² _k of the input signal sample X _k [n]. scaling·
The factor is calculated using the power estimate of the signal derived by the power estimation block 22. However, the hardware required to perform this operation becomes complicated. In order to avoid this complexity, the denormalized variance P _k may be obtained instead of the estimated variance σ ^ ² _k . That is, P _k = 2Nσ ^ ² _k . Therefore, the power estimation block 22 includes the estimation means 2 for obtaining the power estimation value.
21 and a filter means 222 for avoiding sensitive fluctuations, that is, a 1-tap low-pass filter. The hardware is complicated to obtain the power by calculation, but the complexity can be further reduced by reusing the functional blocks already mounted to estimate the power. In a CDMA-based communication system, it is important to minimize the total interference between sectors and base stations.
Means are provided for estimating the total received power at the antenna for each sector and means for controlling and minimizing the output power of the mobiles within the sector with this estimate. The measures required for good system performance are:
Further, the circuit configuration and method according to the present invention can be combined, and in this case, the estimation means 221 is replaced.

3a and 3b are exemplary logarithmic representations of the function for assigning an appropriate scaling factor α _k to the calculated variance level P _k for a signal sample X _k [n]. Is. Said function is applied to said means 211. Thereby, the discrete scaling factor according to the constant interval of the estimated dispersion level P _k is determined, and the scaling factor is assigned. The graph shown in FIG. 3a represents a first method of assigning scaling factors based on the variance values calculated within a fixed interval. Given that the common target signal power level is chosen to be higher than the power level of the received signal, it will be clear that as the said received power level gets lower, the scaling factor has to be chosen higher.

Since the power estimate is noise sensitive, the assigned scaling factor is
Frequent fluctuations between the two values can occur at the boundaries of two adjacent power intervals, thus causing unwanted oscillations in the scaled signal. Therefore, according to another preferred solution, as illustrated in FIG.
The scaling factor assignment may include relay control to prevent oscillating scaling factor assignments at boundaries between power intervals. The power jump value for changing the scaling factor varies depending on which side the power value approaches the normal jump value, ie the boundary between two adjacent power intervals. To decrease the power value, the jump value should be slightly lower than the normal jump value, while to increase the power value, the jump value should be slightly higher than the normal jump value. . Therefore, within a narrow power interval near the boundary, the scaling factor change is delayed and
Oscillation between two scaling factors can be prevented. The unresponsiveness to power fluctuations depends on the width of the narrow power intervals.

FIG. 4 shows a flow chart illustrating the main steps of the method according to the invention, in which the input signal sequence X _k [n] input in block 41 is fed to a common and constant target dispersion level σ ² _ref. It indicates that it is converted to. In the first step, block 42, the received signal is adjusted, if necessary, to have an average value of zero. This adjustment enables scaling processing by multiplication. Then, in block 43, a power estimate P _k is calculated for each signal sample X _k [n], and in block 44 the scaling factor α _k and the target power are dependent on the calculated power estimate P _k. Determine the level σ ² _ref . And
At block 45, each of the received signal samples X _k [n] is scaled to a signal Y _k [n] = α _k−1 · X _k [n] at the target power level, and at block 46, Transferred to demodulator. In an optional step, block 47, if the power estimate is above a threshold, the estimated power level P _k of the AGC unit could be applied to attenuate the analog received signal.

From the above, it is clear that the scaled signal suffers loss due to signal error. This error firstly comes from the fact that the scaling factor is updated only with a common time interval, and secondly, the scaling factor is quantized for distinct intervals of the power estimate of the signal. And preferably from the fact that it is quantized to a value expressed as the sum of two “powers”. This assumes that subsequent units, such as demodulators, scale the signal samples to a common dispersion level. Therefore, it receives and processes signals that deviate significantly from the common dispersion level.
The first solution is to update the scaling factor faster. This will increase the complexity of the demodulators in the subsequent stages. However,
According to an alternative preferred embodiment of the invention, the scaling block 21 of the digital AGC unit according to FIG. 2 produces side information 26 on the quality of scaling and quantization, which side information updates faster than the scaling factor. And is transferred to the demodulator 11 together with the scaled sample value. This additional information 26 is typically a control error of the AGC algorithm, which is derived from the quantization of the scaling factor. By using the above-mentioned additional information, the demodulator can instantaneously and easily switch to the antenna receiving the best input signal, for example, when performing a softer handover.

[Brief description of drawings]

For a better understanding, reference is made to the preferred embodiments of the present invention and the following figures.

FIG. 1 is a diagram showing an overview of a base station receiver, in particular, an uplink signal path input to a base station demodulator.

FIG. 2 is a more detailed view of the signal processing means for one of the signal input paths according to the preferred embodiment of the present invention.

FIG. 3a

3a and 3b are examples of two allocation functions that emphasize the relationship between the power estimate for a signal sample and the scaling factor required to transform the signal sample to some target power level. It is the figure which showed.

FIG. 4 is a flow chart illustrating the main steps of a method according to a preferred embodiment of the present invention.

[Procedure amendment]

[Submission date] March 11, 2002 (2002.3.11)

[Procedure Amendment 1]

[Document name to be amended] Statement

[Name of item to be amended] Title of invention

[Correction method] Change

[Contents of correction]

Title: Circuit used in communication system and receiver unit

─────────────────────────────────────────────────── ─── Continued front page    (81) Designated countries EP (AT, BE, CH, CY, DE, DK, ES, FI, FR, GB, GR, IE, I T, LU, MC, NL, PT, SE), OA (BF, BJ , CF, CG, CI, CM, GA, GN, GW, ML, MR, NE, SN, TD, TG), AP (GH, GM, K E, LS, MW, MZ, SD, SL, SZ, TZ, UG , ZW), EA (AM, AZ, BY, KG, KZ, MD, RU, TJ, TM), AE, AG, AL, AM, AT, AU, AZ, BA, BB, BG, BR, BY, BZ, C A, CH, CN, CR, CU, CZ, DE, DK, DM , DZ, EE, ES, FI, GB, GD, GE, GH, GM, HR, HU, ID, IL, IN, IS, JP, K E, KG, KP, KR, KZ, LC, LK, LR, LS , LT, LU, LV, MA, MD, MG, MK, MN, MW, MX, MZ, NO, NZ, PL, PT, RO, R U, SD, SE, SG, SI, SK, SL, TJ, TM , TR, TT, TZ, UA, UG, UZ, VN, YU, ZA, ZW (72) Inventor Lipstrand, Torbiern             Solna S-169 37, Sweden               Neckro Swagen 43 (72) Inventor Estman, Thomas             Stockholm, Sweden S             115 59, Elegrundsgatan 6 F-term (reference) 5K022 EE01 EE31                 5K067 AA42 CC10 EE02 EE10 GG08                       GG11

Claims (19)

[Claims] 1. A circuit in a receiver unit (10) of a communication system comprising at least one signal input for receiving a signal from a transmitter of said communication system, said signal being a long-term dynamic signal. Having a short range dynamic range which is essentially smaller than the range, the receiver unit (10) comprises a demodulation means (11) for demodulating the received signal; A separate input path to the demodulation means (11), means (13) for converting the received signal into a baseband signal, and means for converting the baseband signal into a digital signal sample. At least a first signal processing means (12, 13) provided in each of the input paths, and a corresponding signal power representing only the dynamic range of the short section. Second signal processing means (16, 20) provided in at least one of said input paths for adapting said signal samples to adapt to said power level and transferring said signal samples to said demodulation means (11). ), And a circuit including. 2. The second signal processing means (16, 20) includes at least a digital automatic gain controller (AGC) unit (161) and a subsequent quantization unit (162). The circuit according to claim 1. 3. The digital AGC unit (161) comprises first means (211) for determining and assigning a scaling factor from a power estimate for each of the digital signal samples, and the assigned scaling factor. Second means (212) for scaling the digital signal samples to a common power level by using. 4. The digital AGC unit (161) according to claim 2, characterized in that it comprises means (26) for directly supplying the demodulation means (11) with additional information concerning the scaling quality. The circuit described. 5. The second processing means (16, 20) means for determining a power estimate of the digital signal sample (221) and means for low pass filtering the power estimate (222).
The circuit of claim 2, including: 6. Circuit according to claim 2, characterized in that said second processing means (16, 20) adjust the reception sequence of the digital signal samples such that the average value is zero. 7. The second processing means (16) comprises an attenuator in the form of an analog AGC unit (163), the attenuator being controlled in response to a power estimate of the digital signal sample. The circuit according to claim 2, wherein: 8. A communication system comprising a plurality of transmitters and a receiver unit, wherein at least one receiver unit comprises the circuit according to any one of claims 1 to 5. A communication system characterized by. 9. The communication system according to claim 8, wherein the at least one receiver unit is part of a radio base station. 10. The communication system according to claim 8 or 9, wherein the communication system is a CDMA-based communication system. 11. Each of the radio base stations includes an essential functional block for estimating signal power of an uplink received signal, and the essential functional block for estimating the signal power is: The communication system according to claim 10, wherein the communication system is reused to extract a signal power estimation value to the digital AGC unit. 12. A method of receiving a signal from a transmitter of a communication system by at least one signal input in a receiver unit of a communication system, the method converting the received signal into a baseband signal. Processing the received signal by converting the baseband signal into digital signal samples; a common and constant power level for the at least one signal input to reduce dynamic range. Transforming the digital signal sample into a signal having a signal, quantizing the transformed signal sample into a signal having a reduced number of bits, and transforming the transformed quantized signal sample into a demodulator. And a step of transferring. 13. Transferring the digital signal samples, determining a scaling factor from a power estimate of the signal sample, and quantizing the scaling factor according to individual intervals of the power estimate. And scaling by multiplying said quantized scaling factor. 14. Transferring said digital signal samples, determining a scaling factor from the power estimates of said signal samples, and scaling said scaling factor to a power of two according to individual intervals of the power estimates. Quantizing so that the sum can be expressed as a value, determining a shift value expressed by using the sum term as an exponent, shifting the signal sample to the left according to the shift value, and shifting the signal sample. 13. The method of claim 12 including the step of scaling by adding the portions that have been added. 15. Method according to claim 13 or 14, characterized in that one scaling factor is determined per signal sample. 16. Method according to any one of claims 12 to 15, characterized in that additional information regarding the scaling quality is provided directly to the demodulator (11). 17. Assigning the quantized scaling factors so that they overlap,
17. Method according to any one of claims 12 to 16, characterized by increasing or decreasing the power estimates at the boundaries of the individual power intervals. 18. The received sequence of digital signal samples is adjusted to have an average value of zero by subtracting an average estimate calculated from the signal samples. 18. The method according to claim 17. 19. The received signal in the analog section of the receiver is attenuated according to the power estimation value of the signal sample of the received signal in the digital section. The method described in.