38 results about "Single tone" patented technology

The embodiment of the invention provides a multi-tone signal sending method and device. The method comprises the steps that a first multi-tone signal is generated, the first multi-tone signal comprises N single-tone signals, the amplitude value of each single-tone signal is the same, the initial phases of at least two single-tone signals in the N single-tone signals are the same, or the absolute values of the initial phases of at least two single-tone signals in the N single-tone signals are the same, and N is an integer larger than or equal to 2; and sending the first multi-tone signal. According to the method disclosed by the invention, the average power of multi-tone signal transmission can be improved, and the transmission distance or the coverage range of the multi-tone signal can be further improved.

The application discloses a piano music segmentation and note recognition algorithm based on a deep learning network, and comprises the following steps: based on a recurrent neural network, obtaining the starting point of each single tone from a continuous piano audio; according to the starting point of each single tone, segmenting a complete piano audio signal into several single tone audios; constructing an audio data set for pitch recognition; pre-processing signals in the audio data set, drawing a transformed spectrogram, and simultaneously segmenting the spectrogram according to the frequency band of each fundamental frequency to obtain a sub-spectrogram; taking the multiple sub-spectrograms segmented from the spectrogram as input, and recognizing the pitch through multiple convolutional neural networks. Compared with a traditional time domain method, the application no longer depends on the selection of a threshold, improves the generalization ability, compared with a traditional frequency domain method, simplifies the calculation process, and improves the running speed.

The invention provides a zero intermediate frequency transceiver receiving channel IQ imbalance calibration method, and relates to the field of radio frequency transceiver calibration. In the method, a single-tone calibration signal generator generates a single-tone signal in a self-calibration process; the dual-tone excitation signal synthesizer is used for generating dual-tone excitation signals in the self-calibration process; the radio frequency switch is communicated with the sending channel and the receiving channel in the calibration process, and an excitation signal sent by the sending channel is looped back to the receiving channel; the error extraction unit is used for extracting an IQ error and calculating a compensation coefficient; and the error compensation unit is used for compensating the link signal. According to the method, a dual-tone test signal is generated by a sending channel, and IQ unbalance is extracted and compensated in a receiving channel according to an algorithm provided by the invention. According to the method, a calibration environment does not need to be set up by external equipment, self-calibration of the chip can be realized, and the calibration cost of a single chip can be effectively reduced.

Embodiments of the present disclosure disclose a method and device for audio quality detection, electronic equipment, medium and product. The method comprises: collecting initial audio data and determining an effective signal segment of the initial audio data; performing slice processing on the initial audio data based on the effective signal segment to obtain a target audio slice that meets a preset slice condition; wherein the preset slice condition is an audio slice that meets a preset audio slice duration and the effective signal segment is not cut and allocated to different target audio slices; processing the target audio slice through a multi-model cascade decision mechanism to obtain a quality detection result of the initial audio data; wherein the multi-model cascade decision mechanism comprises a trained single-tone detection model and a trained audio score model.

A compressor near stall state voiceprint continuous monitoring method, system, medium and equipment, in the method, a predetermined number of microphone measuring points are randomly selected at the inlet to form a microphone array, and multi-channel sound pressure time domain signals are synchronously collected; a rectangular window is added to the multi-channel sound pressure time domain signal to construct an acoustic time domain signal segment, and a fast Fourier transform is performed on the multi-channel acoustic time domain signal segment in the current time domain window to form multiple sound pressure spectra; it is judged whether there are multiple non-synchronous single-tone characteristic frequencies which are non-integral multiples of the current rotation frequency in the sound pressure spectrum; a multi-channel microphone continuous frequency domain observation matrix of the non-synchronous single-tone characteristic frequency is constructed; an orthogonal Fourier transfer matrix is constructed based on the installation azimuth angle of the microphone sensor and the highest order of the mode; a compressor near stall characteristic mode reconstruction model under non-uniform few measuring points is established; the amplitude is solved based on the block orthogonal matching pursuit algorithm, and the compressor near stall working condition monitoring based on the voiceprint characteristics is realized.

The utility model discloses a sound production unit, which comprises a diaphragm assembly, a voice coil, a spacer, two basins and two magnetic circuit assemblies, and at least one of the basins is provided with a sound outlet. The magnetic circuit assemblies are connected with the basins in a one-to-one correspondence manner, and each magnetic circuit assembly is provided with a magnetic gap; the two sides of the vibrating diaphragm assembly are respectively connected with the basin stand, and the center of the vibrating diaphragm assembly is provided with a hollow window. The voice coil is located at the hollow window and is connected with the diaphragm assembly, and the top end and the bottom end of the voice coil respectively extend into the two magnetic gaps; the spacer is arranged in the hollow area of the voice coil, and the edge of the spacer is connected with the inner peripheral wall of the voice coil. The sound production unit is novel in structure, adopts a double-magnetic-circuit design, can fully utilize the front and rear cavity spaces of the sound production unit, increases the total volume of the magnetic steel, and improves the performance of the sound production unit; the structure of the single voice coil enables the sounding unit to be simpler in structure, facilitates the production of the sounding unit, and facilitates the control of the manufacturing cost of the sounding unit.

The application relates to the technical field of vehicle detection, and discloses a tire positioning method based on Bluetooth, a positioning system composed of a center node and four positioning anchor points, which comprises the following steps: selecting one anchor point as a target positioning anchor point to send a broadcast frame, the center node scans the broadcast frame to complete BLE connection; the center node initiates a phase ranging request instruction, sends and receives a single-tone signal in a frequency hopping mode at a predetermined frequency point, calculates a distance value between the center node and the target positioning anchor point according to the channel response of the single-tone signal; the position of the target positioning anchor point is calculated according to the distance value and the position of the center node; other anchor points are selected as target positioning anchor points to complete positioning according to the above steps, and tire positioning is completed in combination with the position of the center node. The positioning system composed of the center node and the four positioning anchor points is used to complete wheel positioning, and high positioning precision and a more convenient and automatic positioning mode can be provided on a Bluetooth device with low power consumption and low cost.

The embodiment of the invention provides a single-tone interference processing method and system, and relates to the technical field of wireless communication. The method comprises the following steps: converting symbols of a time domain into a plurality of subcarriers of a frequency domain; determining a first maximum power value of the subcarrier among the plurality of symbols so as to record the position of the subcarrier corresponding to the first maximum power value; determining an interference subcarrier in the subcarriers based on a counter, and recording the power of the interference subcarrier and the position of the interference subcarrier; and under the condition that the direct-current interference and the interference subcarrier of the transmitting end and the residual direct-current interference of the receiving end exist, processing the direct-current interference and the interference subcarrier of the transmitting end and the residual direct-current interference of the receiving end in sequence based on a preset priority. Through one embodiment of the invention, the problem of low processing efficiency of the single-tone interference in a complex scene in the related technology is at least solved, and the effect of improving the processing efficiency of the single-tone interference in the complex scene is further realized.

A single monotonic arm provided with an arm cover comprises a single monotonic arm body, the interior of a frame connecting section of the single monotonic arm body is hollowed out to form an inner cavity, a first partition plate enabling the frame connecting section to form two assembly stations is arranged in the inner cavity, and a positioning plate is arranged between the first partition plate and the frame connecting section. An arm cover covering one of the assembling stations is arranged between the positioning plate and the first partition plate, a locking assembly capable of retracting under extrusion is arranged in the arm cover, the locking assembly enters the inner cavity when the arm cover covers the assembling stations, and the locking assembly is allowed to retract to the width matched with the inner cavity through extrusion force after entering the locking assembly. Therefore, the transition fit with tightness is realized.

The present disclosure provides a synchronization method involving operation at a timing source and at remote timing circuits respectively communicatively coupled to the timing source. Generating, at the timing source, a reference signal having a reference frequency; generating a multi-tone waveform; and combining the reference signal with the waveform to generate a synchronization signal having a first temporal portion, the first temporal portion comprising the reference signal without the waveform, alternating with a second temporal portion comprising the waveform. The synchronization signal is transmitted to the timing circuit through each of a plurality of signal paths having different respective propagation delays. Each of the timing circuits determines a phase difference between the received single tones to determine a propagation delay of the connected signal path. Each reference signal is individually delayed at the timing circuit based at least on the propagation delay such that the timing circuit provides phase-aligned output reference signals.

The invention relates to the acoustic imaging technology, and provides a single sound source positioning method and device, an acoustic imager and a computer storage medium, the method and device are applied to the acoustic imager, the acoustic imager comprises a microphone array, and the method comprises the following steps: obtaining a plurality of reference sound source drop points at different moments, the reference sound source drop point at each moment represents the maximum sound intensity point of sound from a plurality of sound sources acquired by the microphone array at each moment; according to the positions of all the reference sound source drop points in a preset sound intensity distribution matrix, a target area with the highest distribution density of the reference sound source drop points is determined, and each element in the preset sound intensity distribution matrix corresponds to one sound source drop point where sound can reach; and positioning the target sound source according to the sound source drop point in the target area. According to the invention, stable and reliable single sound source positioning can be realized, and the sound source visualization effect is improved.

The application discloses a musical instrument tone color conversion method based on time-frequency feature refinement, which comprises the following steps: tone color feature extraction is performed on a musical instrument audio data set to establish a target tone color feature library; according to the time domain feature of the target musical instrument tone color, base frequency adjustment, tone length modification, time domain amplitude modulation and pitch synchronous interpolation are performed on the single tone audio to be converted to realize time domain feature conversion; according to the frequency domain feature of the target musical instrument tone color, the harmonic energy ratio and the fractional harmonic energy ratio of the single tone audio after the time domain feature conversion are adjusted to realize frequency domain feature conversion; the single tone audio with the target musical instrument tone color can be converted through the time-frequency feature conversion; and tone color conversion is performed on all single tone audios of the target musical instrument. The application realizes a target musical instrument tone color conversion method capable of extracting transition tone color.

The embodiment of the invention provides a frequency offset determination method and device, a storage medium and electronic equipment, and the method comprises the steps: carrying out the frequency mixing of a first single-tone signal through employing a local oscillator signal in response to a received first single-tone signal, obtaining a first frequency mixing signal, carrying out the signal processing and sampling processing of the first frequency mixing signal, and obtaining a second frequency mixing signal; the method comprises the steps of obtaining a first single-tone signal, obtaining a first sampling signal corresponding to the first single-tone signal, carrying out specified feature point statistics on the first sampling signal to determine initial frequency offset information of the crystal oscillator, obtaining a second sampling signal, and determining the frequency offset of the crystal oscillator based on the second sampling signal and the initial frequency offset information. Through the frequency offset calibration method and device, the frequency offset calibration of the crystal oscillator is realized quickly and accurately, the problem of low efficiency caused by piece-by-piece traversal configuration of a frequency offset determination method in the related technology is solved, and the frequency offset determination efficiency is improved.

The present application relates to a kind of LFMs based on amplitude-phase response measurement timing method, belong to telemetry, radar and test measurement technical field, in the measurement method of amplitude-phase response, it is usually based on the way of single tone or double tone scanning and time-consuming is longer, and single tone cannot measure system phase response;And based on the way of multi-carrier, due to power limit, leading to the signal-to-noise ratio of each sub-carrier reduces, measurement result is poor. Again due to the related advantages of LFM signal, such as bandwidth, energy concentration and pulse compression etc., it gradually obtains application in this kind of measurement calibration system. After using the timing estimation of the present application, not only the amplitude response can be accurately measured, but also the phase response can be accurately measured. The actual measurement results show that the phase measurement curve of each time is very close to even coincide, so the idea of average noise reduction can be further used to improve the measurement accuracy of phase response.

The invention discloses an information metamaterial reflection coefficient measuring method and system based on harmonic detection, and relates to the technical field of electromagnetic material testing. Comprising the following steps: a transmitter generates a single-tone signal, the single-tone signal is incident to an information metamaterial, a periodic control signal for measuring a reflection coefficient of the information metamaterial is loaded to an information metamaterial unit, and the information metamaterial performs time modulation on the incident signal and reflects the incident signal; and the receiver receives the reflection signals carrying the reflection coefficient information of the information metamaterial, and solves reflection coefficient amplitudes and phases corresponding to different control signals according to frequency domain harmonic components of the reflection signals. According to the invention, the periodic control signal is utilized to regulate and control the harmonic component generated by the information metamaterial to measure the reflection coefficient of the information metamaterial, so that the interference influence caused by a background signal generated by environmental scattering is removed; and the measurement of the amplitude and the phase of the reflection coefficient of the information metamaterial can be realized under the condition of any combination of an incident angle and a receiving angle.

A kind of sound-vibration cooperative compressor acoustic resonance monitoring method, system, medium and equipment, method, in different operating conditions, carry out compressor surge test, by short-time Fourier transform to sound pressure time domain signal and vibration time domain signal, the strain signal of corresponding strain gauge is subjected to frequency spectrum analysis and identifies vibration frequency;Joint analysis of the time-frequency characteristics of sound pressure time domain signal and vibration time domain signal, judge whether abnormal non-synchronous peak frequency appears, monitor the difference between vibration abnormal frequency and sound pressure time domain signal abnormal frequency Whether the dominant single tone mode order under abnormal frequency is the same or opposite, by continuous broadband acoustic mode decomposition to sound pressure time domain signal, obtain wave spectrum diagram, monitor whether there is only one vertical bright band in the frequency band below the first-order blade passing frequency in wave spectrum diagram, and the dominant mode order under the frequency is same with the blade disc vibration pitch diameter, then judge that compressor occurs acoustic resonance failure.

The invention discloses a multi-dimensional neural network ADC calibration method and system based on single-tone signal reconstruction, and the method comprises the steps: obtaining a sampling frequency and an input signal frequency which are correspondingly inputted when a trained ADC calibration model based on a multi-dimensional neural network is trained, and obtaining a training signal oversampling rate; obtaining a to-be-calibrated signal oversampling rate corresponding to the to-be-calibrated ADC signal, and comparing the to-be-calibrated signal oversampling rate with the training signal sampling rate; and the ADC signal pair to be calibrated is reconstructed according to a comparison result and a sampling pseudo data interpolation or extraction mode, the reconstructed ADC signal pair is input into the ADC calibration model, pseudo interpolation points are correspondingly removed or extracted signals are re-aligned and spliced after network reasoning is finished, and a calibration result is output. According to the invention, the problem that the multi-dimensional neural network needs to be repeatedly trained in the ADC calibration process due to different signal frequencies or sampling frequencies is avoided.

The application provides a distance zero value self-calibration method based on a radio frequency agile transceiver chip, comprising the following steps: configuring the radio frequency agile transceiver chip as a double-channel mode; selecting a manual gain control mode for a receiving channel gain control mode of the radio frequency agile transceiver chip, and manually setting gain values of a working branch receiving channel and a self-calibration branch receiving channel to be the same; connecting a Tx2 radio frequency output signal of the self-calibration branch transmitting channel to a radio frequency receiving end of a Rx2 receiving channel after the Tx2 radio frequency output signal is subjected to frequency conversion and fixed attenuation processing; adopting a Tone-FM single tone frequency modulation for a transmitting end signal of the self-calibration branch; performing single tone frequency signal demodulation at a receiving end of the self-calibration branch, recovering a single tone signal from the received signal through frequency discrimination technology, and performing phase comparison on the recovered single tone signal and an original single tone signal output by the transmitting end of the self-calibration branch to obtain a distance zero value of the self-calibration branch; and compensating a Tx1 radio frequency output signal of a transmitting channel of a working branch and a Rx1 radio frequency signal of a receiving channel according to the distance zero value obtained by the self-calibration branch.

Various solutions for Internet of Things (IoT) signal transmission with respect to an apparatus, a reader, and an IoT device are described. An apparatus may transmit a carrier wave (CW) to an IoT device. The CW may comprise a single-tone waveform or a multi-tone waveform. The apparatus may receive a backscattered signal from the IoT device. The backscattered signal is generated based on the CW.

A passive communication method, which is used to increase passive tag sensing distance. In the passive communication method, an excitation signal is first sent to a passive tag to energize the passive tag, and a single-tone carrier signal is then sent to the passive tag once energizing of the passive tag is completed, so that the passive tag can reflect the single-tone carrier signal, and thereby send a reflected signal carrying identification information. Energizing a passive tag on the basis of an excitation signal before the passive tag sends a reflected signal by reflecting a single-tone carrier signal, allows the passive tag to have sufficient energy to send the reflected signal, thereby ensuring long-distance transmission of reflected signals sent by passive tags, and effectively increasing passive tag sensing distance.

The utility model relates to the technical field of electronic musical instruments, in particular to an electric blowing pipe which comprises a shell, a string sweeping area is arranged on the outer surface of the shell, and a user can conduct string sweeping operation through the arrangement of the string sweeping area. Wherein the string sweeping area comprises a plurality of strings, and the plurality of strings are arranged at intervals along the width direction of the outer surface of the shell; a circuit board is arranged in the shell, all the sound strings are connected with the circuit board, and different sounds are made by triggering different sound strings; a plurality of strings form a string sweeping area, single-tone playing or string sweeping playing can be carried out, the rich sense of tracks is greatly increased, and under the cooperation of the accompaniment function of the electric blowpipe, the sense of ending can be enhanced by sweeping the strings with the thumb at the end; in addition, a guitar playing and singing mode can be set, playing forms are more diversified, and compared with a common guitar, the electric blowing pipe is smaller in size and convenient to carry. And the electric blowpipe can be connected with various audio devices, so that recording and playing can be easily realized, and a player can conveniently carry out self-evaluation and sharing.

A multi-channel audio signal detection system and implementation method based on a CODEC chip are disclosed. The audio switching unit uses digital communication interface A1 to query whether the current terminal audio unit is in a standby state. If the detected terminal audio unit is in a standby state, an audio loopback channel is established, and a detection permission is sent back to the audio switching unit via digital communication interface B1. Upon receiving the detection permission information, the audio switching unit sends a test single-tone signal to the detected terminal audio unit via audio output interface A1, and simultaneously collects the signal returned from the detected terminal audio unit via audio input interface A1, and analyzes the returned signal. After analysis, the audio switching unit compares the detection result with a preset threshold. If the analysis result is within the preset threshold range, it is considered normal; otherwise, it is marked as a fault, and the detection of subsequent terminal audio units continues. This achieves connectivity detection between audio units in the system.

A passive communication method is applied to increase the sensing distance of a passive tag. In the passive communication method, an excitation signal is sent to a passive tag to charge the passive tag, and a single-tone carrier signal is sent to the passive tag after the passive tag is charged, so that the passive tag can reflect the single-tone carrier signal to send a reflection signal carrying identification information. The passive tag is charged based on the excitation signal and then reflects the single-tone carrier signal to send the reflection signal, so that the passive tag can have enough energy to send the reflection signal, the reflection signal sent by the passive tag can be transmitted for a long distance, and the sensing distance of the passive tag is effectively improved.

The application discloses a single-tone interference estimation method and device, electronic equipment and a storage medium. The method comprises the following steps: performing FFT on a time-domain received signal to obtain a frequency-domain signal, and determining a rough frequency point of a single-tone interference signal in the frequency-domain signal; performing interpolation processing on the rough frequency point and adjacent frequency points to obtain an initial accurate frequency point; taking the initial accurate frequency point as a current accurate frequency point, performing single-point DFT on the current accurate frequency point and adjacent frequency points; performing interpolation processing on the obtained multiple single-point DFT values to obtain a frequency offset, and determining an updated accurate frequency point based on the frequency offset; taking the updated accurate frequency point as a new current accurate frequency point, returning to the step of performing single-point DFT on the current accurate frequency point and adjacent frequency points until a preset iteration termination condition is met, and taking the last updated accurate frequency point as an estimated frequency point of the single-tone interference signal. The scheme significantly improves the frequency estimation accuracy of the single-tone interference, and has low calculation cost.

The invention relates to a phased array self-calibration method and device. The method comprises the following steps of: generating a single-tone signal through a transmitting channel, and simultaneously injecting the single-tone signal into N receiving channels through a power division network; when emission begins, each channel FIFO is reset, and after data are detected, time domain sampling data are stored in an RAM in a mapping mode that a channel number is a high order and a data index is a low order. And when the data index reaches the Fourier transform point number M, stopping collecting. And then for M pieces of sampling data of each channel, only calculating discrete Fourier transform values of single frequency components corresponding to the central frequency point to obtain a frequency domain complex result, and further extracting amplitude values and phase values of each channel at the central frequency point. And finally, selecting one channel as a reference, and calculating relative amplitude compensation values and phase compensation values of other channels for correcting each radio frequency channel in real time and eliminating amplitude-phase inconsistency. By adopting the method, continuous and efficient storage of multi-channel data with low hardware overhead can be realized.

The invention discloses a multi-channel audio signal detection system based on a CODEC chip and an implementation method. An audio switching unit inquires whether a current terminal audio unit is in a waiting state or not through a digital communication interface A1; if the detected terminal audio unit is in a waiting state, an audio loopback channel is established, and allowed detection is sent back to the audio switching unit through a digital communication interface B1; when the audio switching unit receives the detection allowing information, a test single-tone signal is sent to the detected terminal audio unit through an audio output interface A1, and meanwhile, a signal returned from the detected terminal audio unit is collected through an audio input interface A1 and analyzed; after the analysis is completed, the audio switching unit compares the detection result with a preset threshold value, if the analysis result is within the preset threshold value range, the detection result is normal, otherwise, the detection result is marked as a fault, and then the subsequent detection of the terminal audio unit is continued; and connectivity detection among the audio units in the system is realized.

The invention discloses a multifunctional electronic musical instrument which comprises a fingerboard area which is arranged in a multi-string and multi-product matrix mode and comprises a plurality of fingerboard keys, a first tone key area which comprises a plurality of first tone keys which are linearly arranged along a first direction, and a second tone key area which comprises a plurality of second tone keys which are linearly arranged along a second direction, the pronunciation module is used for outputting musical sound; the control module is electrically connected with the fingerboard key, the first tone key and the sounding module respectively; the control module is configured to respond to continuous pressing of the fingerboard key in a preset continuous mode, and control the pronunciation module to output continuous tenuto; and in response to triggering of any first tone key, controlling the pronunciation module to make a single tone with a corresponding pitch. Two-hand cooperation of musical instrument playing is achieved, meanwhile, continuous tenuto of the fingerboard area and single tone of the first tone key area are combined, fingering is simple, the fingerboard area is arranged in a multi-string and multi-product mode, grades and string position pitches are distinguished, the keys are adopted, operation of a player is facilitated, and unification and high efficiency of musical instrument playing are achieved.

The invention discloses a method and a device for generating an electromagnetic interference signal. The method comprises the following steps of: S101, detecting a frequency point, a bandwidth and a frequency hopping period of a target communication signal in real time; s102, target communication signal prior information detected in real time is utilized, aiming type signal styles are automatically and dynamically generated and matched through simulation, and the signal styles capable of being generated comprise single-tone signals, multi-tone signals, narrow-band signals, pulse signals, noise frequency modulation signals, noise amplitude modulation signals, noise phase modulation signals and / or linear frequency sweeping signals; and S103, generating an actual multi-system signal based on a signal generator and a spectrum analyzer, dynamically adjusting parameters under different signal styles, including single-tone, multi-tone, narrow-band, pulse, noise frequency modulation signal frequency points, multi-tone signal spectrum number and interval, narrow-band signal bandwidth, pulse signal interval and duty ratio, and realizing dynamic reconfiguration of the parameters.

The invention belongs to the technical field of non-cooperative signal processing, and particularly relates to a method for calculating MPSK interference performance of single-tone interference. According to the method, under the condition that the signal-to-noise ratio and the interference-to-signal ratio are known, communication signal power is normalized, the power and amplitude of communication signals, interference signals and noise processed by a receiver are calculated, probability distribution of output signals is calculated according to a formula, integration is conducted on a corresponding judgment area demodulated by MPSK, the judgment accuracy is obtained, and the error rate can be obtained. And then, according to the probability distribution of the output signal, carrying out integration on each MPSK judgment region to obtain an error transition probability matrix of the signal, calculating to obtain the entropy of an output sequence and the conditional entropy of the output sequence when the input sequence is known, and obtaining mutual information according to a formula. According to the method, the interference performance can be well estimated when the MPSK signal is interfered by the single-tone interference signal.

An ultrasound computed tomography apparatus includes: an imaging region configured to accommodate a medium to be imaged; an array of ultrasonic elements arranged around the imaging region, the ultrasonic elements comprising a transmitter and a plurality of receivers; a respective receiver circuit for each of the plurality of receivers; and a processing resource configured to perform a computer transmission tomography. The transmitter is configured to transmit a sequence of single-tone continuous wave signals through the imaging region, the sequence of single-tone continuous wave signals including at least one single-tone continuous wave signal having a first frequency and at least one single-tone continuous wave signal having a second, different frequency. Each of the plurality of receivers is configured to receive a sequence of single tone continuous wave signals that have been transmitted through an imaging region. A respective receiver circuit for each of the plurality of receivers is configured to mix each received single-tone continuous wave signal in the sequence of single-tone continuous wave signals with a respective single-frequency mixed signal having the same frequency as the single-tone continuous wave signal, to obtain a respective in-phase value and a respective quadrature value of the single-tone continuous wave signal. The processing circuitry is configured to process the obtained in-phase and quadrature values by computer transmission tomography to obtain an image of the imaged region.