Radio receiver for different data rates

The present invention relates to a radio receiver and to methods of operation thereof. There will be described below, by way of example in illustration of the invention a low-power-consumption radio receiver which is able to receive a plurality of channels at different frequencies and which has a reduced power consumption.

Various radio receivers, such as portable telephones, and mobile computers have been proposed. Recently, a software radio system has been proposed and developed.

This software radio system enables the same device to perform communications at different frequencies, or by employing different communication schemes, as a result of making software changes.

A reduction in power consumption presents a significant challenge for a radio receiver, and more specifically, for a portable radio receiver, and demands have arisen for the development of a lower-power consumption radio receiver.

A feature of an arrangement to be described, by way of example in illustration of the present invention is that it enables a low-power-consumption radio receiver which reduces power consumption to be provided by decreasing the data rate of received data.

In a particular radio receiver to be described by way of example in illustration of the present invention, there is a device for decreasing the data rate of received data in accordance with the channel frequency of the received data, and a device for performing signal processing of the received data whose data rate has been decreased.

There will be described below, by way of example in illustration of the invention, a radio receiver which is capable of receiving a plurality of different channel frequencies, the receiver including an A/D converter for A/D-converting a received signal, a signal processing section for performing digital signal processing of a signal which has undergone digital conversion in the A/D converter, and either a sampling frequency changing device for changing the sampling frequency in the A/D converter in accordance with the frequency of the received signal, or a data rate changing device for changing the data rate of the signal which is input to the signal processing section and which has undergone digital conversion.

A power consumption reducing method for a radio receiver to be described below, by way of example in illustration of the invention includes the steps of decreasing the data rate of received data, and performing signal processing of the received data whose data rate has been decreased.

Power consumption poses a significant challenge to portable telephones, mobile computers, and the like. It is known that power consumption can be reduced in signal processing by decreasing the clock speed used for the processing. In the particular arrangements to be described by way of example in illustration of the present invention, the clock speed required for signal processing is decreased by decreasing the data rate, thereby realizing a power consumption reducing effect.

The following description and drawings disclose, by means of examples, the invention which is characterised in the appended claims, whose terms determine the extent of the protection conferred hereby.

In the drawings: <ul id="ul0001" list-style="none" compact="compact"><li><figref idref="f0001">Fig. 1</figref> is a block schematic diagram showing the main part of a radio receiver,</li><li><figref idref="f0002">Figs. 2A to 2G</figref> are timing charts for explaining a variable sampling frequency decreasing method for use in the receiver shown in <figref idref="f0001">Fig. 1</figref>,</li><li><figref idref="f0003">Figs. 3A to 3G</figref> are timing charts for explaining another variable sampling frequency decreasing method for use in the receiver shown in <figref idref="f0001">Fig. 1</figref>,</li><li><figref idref="f0004">Fig. 4</figref> is a block schematic diagram showing the main part of another radio receiver, and</li><li><figref idref="f0005">Figs. 5A to 5D</figref> are timing charts for explaining a variable sampling frequency decreasing method for use in the receiver shown in <figref idref="f0004">Fig. 4</figref>.</li></ul>

Referring to <figref idref="f0001">Fig. 1</figref>, the radio receiver includes an analog filter 101 for limiting an input signal band, an A/D converter 102 for sampling an input signal and converting it into a digital signal, an oscillator 103 for supplying a clock sampling signal to the A/D converter 102, and a signal processing section 104.

The signal processing section 104 includes a DSP or CPU, a ROM, and a RAM. This section may be implemented by software or hardware by using a logic circuit, such as a gate array or an FPGA.

The signal processing section 104 includes a quadrature demodulation section 105 for quadrature-demodulating digital data 118 received from the A/D converter 102, a first data rate changing section 110 for changing the data rate of I and Q signals after quadrature demodulation, a digital filter section 111 for extracting a channel to be received, a second data rate changing section 112, a demodulation section 113, a next-stage signal processing section 114 for performing conventional signal processing such as decoding, and a control section 115.

The quadrature demodulation section 105 includes mixers 106 and 107, a digital oscillator 109, and a π/2 shift section 108.

The operation of the receiver shown in <figref idref="f0001">Fig. 1</figref> will now be described with reference to <figref idref="f0002">Figs. 2A to 2G</figref>, or <figref idref="f0003">Figs. 3A to 3G</figref>.

Assume that in a radio receiver capable of receiving several different channel frequencies in a receivable frequency band (to be referred to as a receivable band hereinafter) B, a given channel frequency fIFn (n is an arbitrary number) is to be received. The operation of part of the receiving section shown in <figref idref="f0001">Fig. 1</figref> will now be described.

An analog input signal 116 is band-limited by the analog filter 101. The analog filter 101 is a filter that passes the receivable band B.

The band-limited signal 117 which is output from the filter 101 is sampled and quantized by the A/D converter 102 using a sampling clock signal generated by the oscillator 103 and having a sampling frequency fs. According to the Nyquist theorem, the sampling frequency fs must be two times, or more, that of the receivable band B. The data rate of the signal 118 output from the converter 102 is the sampling frequency fs.

The signal processing section 104 includes a DSP or CPU, a ROM, and a RAM when it has a software configuration. When this section is to have a logic configuration, the same functions are implemented by a gate array, FPGA, or the like.

The sampled signal 118 (data rate fs) which is output from the A/D converter 102 is input to the signal processing section 104.

To quadrature-demodulate the received channel frequency fIFn, the control section 115 controls the quadrature demodulation section 105 constituted by the mixers 106 and 107, the digital oscillator 109, and the π/2 shift section 108. More specifically, the control section 115 controls the digital oscillator 109 to set the channel frequency to fIFn.

I and Q signals 119 and 120, after quadrature demodulation, are input to the first data rate changing section 110. The first data rate changing section 110 performs a data rate changing operation on the basis of the information of the received channel fIFn under the control of the control section 115, thereby setting the data rate of output signals 121 and 122 to fsn (n is an arbitrary number).

Signals 123 and 124 which are output from the digital filter section 111 serving as a channel filter are input to the second data rate changing section 112. The control section 115 controls the second data rate changing section 112 so that output signals 125 and 126 from the second data rate changing section 112 are set to a data rate fsaO. At this time, the data rate fsaO must satisfy the Nyquist condition with respect to a pass band B0 required for the reception. In this case, since the signals 125 and 126 have been decreased to the baseband, fas0 is set under the condition that the data rate should be the pass band B0 or more.

These signals 125 and 126 are input to the demodulation section 113 to be demodulated. A demodulated signal 127 is transferred to the next-stage signal processing section 114. The next-stage signal processing section 114 is an existing signal processing section and has constituent elements for control by signal formats, decoding, speech reproduction, and the like. The next-stage signal processing section 114 outputs sound, light, speech, FAX data, LCD data, PC data, and the like.

The data rate change operation in the second data rate changing section 112 may be performed in the digital filter section 111.

A variable sampling frequency decreasing method employed with the arrangement shown in <figref idref="f0001">Fig. 1</figref> will now be described with reference to <figref idref="f0002">Figs. 2A to 2G</figref>.

<figref idref="f0002">Fig. 2A</figref> shows the characteristics of an analog filter, including the receivable band B in its pass band and the reception channel frequencies fIF1 and fIF2, in the radio receiver of <figref idref="f0001">Fig. 1</figref>.

In addition, a reference symbol Ba denotes the attenuation band of the analog filter 101, which is set to a condition sufficient for cutting off interference by other channels, or noise, and to allow sufficient attenuation at frequencies that differ by a certain value or more. In this case, "sufficient attenuation" indicates a level at which aliasing noise (to be described later) is not superimposed on the necessary received signal sufficiently to degrade its sensitivity.

The received channel frequencies are the channel frequencies fIF1 and fIF2, and a larger frequency difference between the frequency at which a sufficient attenuation band is obtained and the channel frequency fIF1 is denoted by a reference symbol B1. <figref idref="f0002">Fig. 2A</figref> also shows the sampling frequency fs in an A/D conversion in which the frequency must be two times, or more, the receivable band B, according to the Nyquist theorem.

<figref idref="f0002">Fig. 2B</figref> shows a case in which the receiving channel for the frequency fIF1 is decreased to the baseband in the quadrature demodulation section 105. Since both the I and Q signals 119 and 120 undergo the same frequency change, changes in the frequencies of these signals are not separately shown.

<figref idref="f0002">Fig. 2C</figref> shows the signals 121 and 122 obtained when the data rate is changed to fs1 by the first data rate changing section 110. Aliasing noise is a general definition. In digital signal processing, a frequency 1/2 or more of the sampling frequency which falls out of the Nyquist definition becomes aliasing noise. Referring to <figref idref="f0002">Fig. 2C</figref>, fs1 must be set to prevent aliasing noise from entering a pass band B0 required for reception. In this case, since the frequency separation B1 from the attenuation region shown in <figref idref="f0002">Figs. 2A and 2B</figref> becomes dominant, fs1 must satisfy the condition given by: <maths id="math0001" num="(1)"><math display="block"><mi>fs</mi><mo>⁢</mo><mn mathvariant="normal">1</mn><mo mathvariant="normal">≧</mo><mi mathvariant="normal">B</mi><mo>⁢</mo><mn mathvariant="normal">1</mn><mo mathvariant="normal">+</mo><mi mathvariant="normal">B</mi><mo>⁢</mo><mn mathvariant="normal">0</mn><mo mathvariant="normal">/</mo><mn mathvariant="normal">2</mn></math><img file="EP2249531A2_D0001.tif" /></maths> where B0 is the required pass band.

<figref idref="f0002">Fig. 2D</figref> shows the case in which the required pass band B0 is extracted by the digital filter section 111, and the data rate is changed to fsa0 by the second data rate changing section 112.

Since the signals have already been band-limited by the digital filter section 111, even if the sampling rate is decreased to fsa0, two times, or more, the pass band B0, aliasing noise exerts no influence on the signals. These signals are the signals 125 and 126, which are sent to the demodulation section 113.

Reception of the channel frequency fIF2 in <figref idref="f0002">Fig. 2E</figref> will be described next. Assume that a band with a larger frequency separation from the attenuation frequency of the analog filter 101 is represented by B2 as in the case of the receiving channel frequency fIF1.

When the frequency is changed to the baseband frequency by quadrature demodulation, and the sampling frequency is decreased to fs2, as shown in <figref idref="f0002">Fig. 2F</figref>, aliasing noise is produced as shown in <figref idref="f0002">Fig. 2F</figref>. To prevent this aliasing noise from entering the required frequency band B0 of the reception channel frequency fIF2, the sampling frequency fs2 must satisfy the condition given: <maths id="math0002" num="(2)"><math display="block"><mi>fs</mi><mo>⁢</mo><mn mathvariant="normal">2</mn><mo mathvariant="normal">≧</mo><mi mathvariant="normal">B</mi><mo>⁢</mo><mn mathvariant="normal">2</mn><mo mathvariant="normal">+</mo><mi mathvariant="normal">B</mi><mo>⁢</mo><mn mathvariant="normal">0</mn><mo mathvariant="normal">/</mo><mn mathvariant="normal">2</mn></math><img file="EP2249531A2_D0002.tif" /></maths> where B0 is the required pass band.

<figref idref="f0002">Fig. 2G</figref> shows a case in which the pass band B0 is extracted by the digital filter section 111, and the data rate is changed to fsa0 by the second data rate changing section 112.

Since the signals have already been band-limited by the digital filter section 111, even if the sampling rate is decreased to fsa0, aliasing noise exerts no influence on the signals. These signals are the signals 125 and 126, which are sent to the demodulation section 113.

As is obvious from these results, the data rates fs1 and fs2 of the data 121 and 122 input to the digital filter section 111 can be effectively decreased on the basis of the respective received frequencies fIF1 and fIF2.

Another variable sampling frequency decreasing method in the embodiment shown in <figref idref="f0001">Fig. 1</figref> will be described with reference to <figref idref="f0003">Figs. 3A to 3G</figref>.

<figref idref="f0003">Fig. 3A</figref> shows an analog filter band including the receivable band B and the sampling frequency fs in A/D conversion which is two or more times the receivable band B. In this case, a channel frequency fIF3 to be received is at the centre of the sufficient attenuation band Ba (≧ receivable band B) of the analog filter, whereas a channel frequency fIF4 is the lowest channel at the end of the receivable band B.

<figref idref="f0003">Fig. 3B</figref> shows the I and Q signals 119 and 120 obtained when the receiving channel frequency fIF3 is decreased to the baseband in the quadrature demodulation section 105.

<figref idref="f0003">Fig. 3C</figref> shows the signals 121 and 122 obtained when the data rate is changed to fs3 in the first data rate changing section 110. That is, in the case shown in <figref idref="f0003">Fig. 3C</figref>, to set fs3 so as to prevent aliasing noise from entering the required pass band B0, fs3 must satisfy the condition given by: <maths id="math0003" num="(3)"><math display="block"><mi>fs</mi><mo>⁢</mo><mn mathvariant="normal">3</mn><mo mathvariant="normal">≧</mo><mfenced separators=""><mi>Ba</mi><mo mathvariant="normal">+</mo><mi mathvariant="normal">B</mi><mo>⁢</mo><mn mathvariant="normal">0</mn></mfenced><mo mathvariant="normal">/</mo><mn mathvariant="normal">2</mn></math><img file="EP2249531A2_D0003.tif" /></maths> where Ba is the band with which the analog filter cannot satisfactorily cut off noise, and B0 is the required pass band.

<figref idref="f0003">Fig. 3D</figref> shows the case in which the required pass band B0 is extracted by the digital filter section 111, and the data rate is changed to fsa0 by the second data rate changing section 112.

Since the signals have already been band-limited by the digital filter section 111, and fsa0 is higher than the pass band B0, even if the sampling rate is decreased to fsa0, no influences are exerted on the signals. These signals are the signals 125 and 126, which are sent to the demodulation section 113.

The reception of a channel frequency fIF4 shown in <figref idref="f0003">Fig. 3E</figref> will be described next.

<figref idref="f0003">Fig. 3F</figref> shows the signals 121 and 122 obtained when the receiving channel frequency fIF4 is decreased to the baseband in the quadrature demodulation section 105, and the data rate is changed to fs4 in the first data rate changing section 110. As in the case shown in <figref idref="f0003">Fig. 3C, Fig. 3F</figref> shows aliasing noise. That is, in the case shown in <figref idref="f0003">Fig. 3F</figref>, fs4 is set under the condition given below to prevent aliasing noise from entering the required band B0: <maths id="math0004" num="(4)"><math display="block"><mi>fs</mi><mo>⁢</mo><mn mathvariant="normal">4</mn><mo mathvariant="normal">≧</mo><mfenced separators=""><mi>Ba</mi><mo mathvariant="normal">+</mo><mi mathvariant="normal">B</mi></mfenced><mo mathvariant="normal">/</mo><mn mathvariant="normal">2</mn></math><img file="EP2249531A2_D0004.tif" /></maths> where Ba is the band with which the analog filter cannot satisfactorily cut off noise, and B is the receivable band described above.

<figref idref="f0003">Fig. 3G</figref> shows the case in which a required band is extracted by the digital filter section 111, and the data rate is changed to fsa0 by the second data rate changing section 112.

Since the signals have already been band-limited by the digital filter section 111, even if the sampling rate is decreased to fsa0, no influences are exerted on the signals. These signals are the signals 125 and 126, which are sent to the demodulation section 113.

As is obvious from these results, the sampling frequencies fs3 and fs4 of data input to the digital filter section 111 can be changed on the basis of the different receiving channel frequencies fIF3 and fIF3, and the operating frequency of the digital filter can be decreased independently. In this case, the data rates fs3 and fs4 based on the receiving channel frequencies fIF3 and fIF4 represent conditions for the minimum and maximum sampling frequencies. In addition, with regard to the condition of the receiving channel frequency fIF4 on which the sampling condition is maximized, there is no difference between the lowest channel frequency and the highest channel frequency in the receivable band B.

Referring to <figref idref="f0004">Fig. 4</figref>, there is shown a radio receiver which includes an analog filter 401 for band-limiting an input signal, an A/D converter 402 for sampling an input signal and converting it into a digital signal, an oscillator 403 for supplying a sampling clock to the A/D converter 402, and a signal processing section 404.

The signal processing section 404 includes a DSP or CPU, a ROM, and a RAM. This section may be implemented by software or hardware by using a logic circuit such as a gate array or FPGA.

The signal processing section 404 includes a down conversion section 405 for down-converting digital data 416 received from the A/D converter 402, a first data rate changing section 408 for changing the data rate of a signal which has undergone down-conversion, a digital filter section 409 for extracting a channel to be received, a second data rate changing section 410, a demodulation section 411, a next-stage signal processing section 412 for performing conventional signal processing such as decoding, and a control section 413.

The down conversion section 405 includes a mixer 406 and a digital oscillator 407.

<figref idref="f0005">Figs. 5A to 5D</figref> are timing charts for use in explaining a variable sampling frequency decreasing method used in the receiver shown in <figref idref="f0004">Fig. 4</figref>.

With reference to <figref idref="f0004">Fig. 4</figref> and <figref idref="f0005">Figs. 5A to 5D</figref>, assume that in a radio receiver capable of receiving several different channel frequencies in a receivable band B, a given channel frequency fIFn (n is an arbitrary number) is to be received. The operation to be carried out in part of the receiving section will now be described.

An analog input signal 414 is band-limited by the analog filter 401. This analog filter 401 is a filter that passes the receivable band B.

A band-limited signal 415 is sampled and quantized by the A/D converter 402 with a sampling clock generated by the oscillator 403 and having a sampling frequency fs. According to the Nyquist theorem, this sampling frequency fs needs to be two or more times the receivable band B. In this case, the data rate of a signal 416 is the sampling frequency fs.

The signal processing section 404 includes a DSP or CPU, a ROM, and a RAM, when it has a software configuration. When this section is to have a logic configuration, the same function is implemented by a gate array, FPGA, or the like.

The signal 416 (data rate fs) sampled by the A/D converter 402 is input to the signal processing section 404.

First of all, the control section 413 controls the down conversion section 405 constituted by the mixer 406 and the digital oscillator 407 to down-convert the received channel frequency fIFn. More specifically, the control section 413 controls the digital oscillator 407 to set the frequency to fIFn-fIF0 or fIFn+fIF0. In this case, the frequency fIF0 is a frequency required for demodulation in the demodulation section 411.

As a result, a signal 417 having undergone down-conversion becomes fIF0. This signal 417 is input to the first data rate changing section 408. The control section 413 changes the data rate on the basis of the information of the received frequency fIFn. As a consequence, the data rate of an output signal 418 is set to fsbn (n is an arbitrary number).

A signal 419 having passed through the digital filter section 409 serving as a channel filter is input to the second data rate changing section 410. control section 413 controls the second data rate changing section 410 to make an output signal 420 from the second data rate changing section 410 have a data rate of fsb0. In this case, the data rate fsb0 must satisfy a Nyquist condition with respect to the frequency fIF0 and a pass band B0 required for reception. In this case, the condition is fsb0 ≧ 2 x (fIF0 + B0/2).

The signal 420 is input to the demodulation section 411 to be demodulated. A demodulated signal 421 is transferred to the next-stage signal processing section 412. The next-stage signal processing section 412 is an existing signal processing unit and has constituent elements for control by signal formats, decoding, speech reproduction, and the like. The next-stage signal processing section 412 outputs sound, light, speech, FAX data, LCD data, PC data, and the like.

The data rate change operation in the second data rate changing section 410 may be performed in the digital filter section 409.

A variable sampling frequency decreasing method in the receiver shown in <figref idref="f0004">Fig. 4</figref> will be described with reference to <figref idref="f0005">Figs. 5A to 5D</figref>. <ul id="ul0002" list-style="none" compact="compact"><li><figref idref="f0005">Fig. 5A</figref> shows an analog filter band including the receivable band B, the sampling frequency fs in the A/D conversion which is two or more times the receivable band B according to the Nyquist theorem, and a channel frequency fIF5 to be received.</li><li><figref idref="f0005">Fig. 5B</figref> shows the signal 417 obtained when the frequency is decreased to a frequency fIF0 required to demodulate the receiving channel frequency fIF5 in the down conversion section 405.</li><li><figref idref="f0005">Fig. 5C</figref> shows the signal 418 obtained when the data rate is changed to fs5 by the first data rate changing section 408. Referring to <figref idref="f0005">Fig. 5C</figref>, the data rate fs5 must be set to prevent aliasing noise from entering the pass band B0 required for reception.</li><li><figref idref="f0005">Fig. 5D</figref> shows the case in which the required pass band B0 is extracted by the digital filter section 409, and the data rate is changed to fsb0 by the second data rate changing section 410.</li></ul>

Since the signal has already been band-limited by the digital filter section 409, and the frequency fsb0 satisfies fsb0 ≧ 2 x (fIF0 + B0/2), even if the sampling rate is decreased to fsb0, aliasing noise exerts no influence on the signal. This signal is the signal 420 and supplied to the demodulation section 411.

Since the present invention is based on a device including a plurality of channel frequencies, it most likely uses FDMA. In practice, however, channels for CDMA or TDMA can be included in the simultaneous receivable band.

It will be understood that, although particular arrangements have been described by way of example in illustration of the invention, variations and modifications thereof, as well as other embodiments may be conceived within the scope of the appended claims.