Abstract: The present invention can include a method of call processing using a distributed voice browser including allocating a plurality of service processors configured to interpret parsed voice markup language data and allocating a plurality of voice markup language parsers configured to retrieve and parse voice markup language data representing a telephony service. The plurality of service processors and the plurality of markup language parsers can be registered with one or more session managers. Accordingly, components of received telephony service requests can be distributed to the voice markup language parsers and the parsed voice markup language data can be distributed to the service processors.

Abstract: According to the present invention, a method for integrating processes with a multi-faceted human centered interface is provided. The interface is facilitated to implement a hands free, voice driven environment to control processes and applications. A natural language model is used to parse voice initiated commands and data, and to route those voice initiated inputs to the required applications or processes. The use of an intelligent context based parser allows the system to intelligently determine what processes are required to complete a task which is initiated using natural language. A single window environment provides an interface which is comfortable to the user by preventing the occurrence of distracting windows from appearing. The single window has a plurality of facets which allow distinct viewing areas. Each facet has an independent process routing its outputs thereto. As other processes are activated, each facet can reshape itself to bring a new process into one of the viewing areas.

Abstract: In one embodiment, at least first and second channels in a frame of the audio signal are independently subdivided into blocks if the first and second channels are not correlated with each other. At least two of the blocks have different block lengths. Furthermore, the first and second channels are correspondingly subdivided into blocks such that the lengths of the blocks into which the second channel is subdivided correspond to the lengths of the blocks into which the first channel is subdivided if the first and second channels are correlated with each other. At least two of the blocks have different block lengths.

Abstract: In some embodiments, a processor-readable medium stores code representing instructions to cause a processor to receive an input signal having a first component and a second component. An estimate of the first component of the input signal is calculated based on an estimate of a pitch of the first component of the input signal. An estimate of the input signal is calculated based on the estimate of the first component of the input signal and an estimate of the second component of the input signal. The estimate of the first component of the input signal is modified based on a scaling function to produce a reconstructed first component of the input signal. The scaling function is a function of at least one of the input signal, the estimate of the first component of the input signal, the estimate of the second component of the input signal, or a residual signal.

Abstract: The present invention is based on the finding that parameters including a first set of parameters of a representation of a first portion of an original signal and including a second set of parameters of a representation of a second portion of the original signal can be efficiently encoded, when the parameters are arranged in a first sequence of tuples and in a second sequence of tuples, wherein the first sequence of tuples comprises tuples of parameters having two parameters from a single portion of the original signal and wherein the second sequence of tuples comprises tuples of parameters having one parameter from the first portion and one parameter from the second portion of the original signal. An efficient encoding can be achieved using a bit estimator to estimate the number of necessary bits to encode the first and the second sequence of tuples, wherein only the sequence of tuples is encoded, that results in the lower number of bits.

Abstract: A microphone apparatus for processing and outputting an output signal of a microphone array including at least nine microphones includes a directivity function processing circuit that converts the output signal of the microphone array into a unidirectional signal and that outputs the unidirectional signal. The directivity function processing circuit expands a directivity function whose variable is an incident angle of an acoustic wave into a Fourier series up to at least third order. The variable in the expanded expression is produced from output signals of the microphones forming the microphone array.

Abstract: A “STAC Codec” provides lossless audio compression and decompression by processing an audio signal using integer-reversible modulated lapped transforms (MLT) to produce transform coefficients. Transform coefficients are then encoded using a backward-adaptive run-length Golomb-Rice (RLGR) encoder to produce losslessly compressed audio signals. In additional embodiments, further compression gains are achieved via an inter-block spectral estimation and data sorting strategy. Further, compression in the transform domain allows the bitstream to be partially decoded, using the corresponding RLGR decoder, to reconstruct the frequency-domain coefficients. These frequency-domain coefficients are then directly used to speed up various transform-domain based applications such as transcoding media to lossy or other formats, search, identification, visualization, watermarking, etc.

Abstract: Methods, systems, and apparatus, including computer programs encoded on a computer storage medium, for processing voice commands. In one aspect, a method includes receiving an audio signal at a server, performing, by the server, speech recognition on the audio signal to identify one or more candidate terms that match one or more portions of the audio signal, identifying one or more possible intended actions for each candidate term, providing information for display on a client device, the information specifying the candidate terms and the actions for each candidate term, receiving from the client device an indication of an action selected by a user, where the action was selected from among the actions included in the provided information, and invoking the action selected by the user.

Abstract: The invention relates to a process for operating a voice dialog system and a voice dialog system which can be controlled over a telecommunications link by a communications terminal, a speech element transmitted by the communications terminal being received by a receiving unit of the voice dialog system and being analyzed for statement content in a processing unit, the speech element being filed in a memory assigned to the processing unit and after the telecommunications link is broken being analyzed by the processing unit.

Abstract: An apparatus in one example has: a receiver configured to receive an input signal in a first encoding format, the input signal having an input payload; and a transcoder operatively coupled to the receiver, the transcoder structured to transcode in a single channel the first encoding format to a second encoding format, the transcoder configured to generate an output signal in the second encoding format based on the input signal, the output signal having an output payload; and wherein the transcoder is configured to switch between providing encrypted data in the output payload and non-encrypted data in the output payload.

Abstract: Provided are a method for emotion communication to share a user's emotions between an emotion signal sensing device and an emotion service providing device.

Abstract: A microphone unit 1 of the present invention includes a case 10 having an internal space 100, a partition member 20 which is provided in the case, and at least partially composed of a vibrating membrane 30, that splits the internal space into a first space 102 and a second space 104, and an electrical signal output circuit 40 that outputs an electrical signal on the basis of vibration of the vibrating membrane. A first through hole 12 through which the first space 102 and an external space of the case are communicated with each other, and a second through hole 14 through which the second space 104 and the external space of the case are communicated with each other are formed in the case 10. In accordance with the present invention, it is possible to provide a high-quality microphone unit whose outer shape is small and which is capable of performing thorough noise cancellation.

Abstract: Storing audio data encoded in any of a plurality of different audio encoding formats is enabled by parametrically defining the underlying format in which the audio data is encoded, in audio format and packet table chunks. A flag can be used to manage storage of the size of the audio data portion of the file, such that premature termination of an audio recording session does not result in an unreadable corrupted file. This capability can be enabled by initially setting the flag to a value that does not correspond to a valid audio data size and that indicates that the last chunk in the file contains the audio data. State information for the audio data, to effectively denote a version of the file, and a dependency indicator for dependent metadata, may be maintained, where the dependency indicator indicates the state of the audio data on which the metadata is dependent.

Abstract: A handheld communication device is used to capture video streams and generate a multiplexed video stream. The handheld communication device has at least two cameras facing in two opposite directions. The handheld communication device receives a first video stream and a second video stream simultaneously from the two cameras. The handheld communication device detects a speech activity of a person captured in the video streams. The speech activity may be detected from direction of sound or lip movement of the person. Based on the detection, the handheld communication device automatically switches between the first video stream and the second video stream to generate a multiplexed video stream. The multiplexed video stream interleaves segments of the first video stream and segments of the second video stream. Other embodiments are also described and claimed.

Abstract: A method of and an apparatus for encoding/decoding an MPEG-4 bit sliced arithmetic coding (BSAC) audio bitstream having ancillary information. A time domain audio signal is converted to a frequency domain audio signal and quantized. A number of data bits is counted and a number of available bits per layer is obtained. The number of available bits per layer is modified considering the size of ancillary information. Actual audio data is encoded in units of layers and ancillary information is embedded in the encoded bitstream. A header is decoded and a layer structure of an audio bitstream is calculated to determine the size of the ancillary information as a difference between a size of data up to a top layer and a size of a frame. The ancillary information is extracted to improve meta data and sound quality of audio contents.

Abstract: Certain embodiments disclosed herein relate to systems and methods for recording audio and video. In particular, in one embodiment, a method of recording audio signals is provided. The method includes recording audio signals with a plurality of distributed audio transducers to create multiple recordings of the audio signals and providing each of the multiple recordings of the audio signals to a computing device. The computing device combines each of the multiple recordings into a master recording and determines a source for each audio signal in the master recording. Additionally, the computing device stores each audio signal in separate audio files according to the determined source of each audio signal.

Abstract: A method and apparatus for implementation of real-time speech recognition using a handheld computing apparatus are provided. The handheld computing apparatus receives an audio signal, such as a user's voice. The handheld computing apparatus ultimately transmits the voice data to a remote or distal computing device with greater processing power and operating a speech recognition software application. The speech recognition software application processes the signal and outputs a set of instructions for implementation either by the computing device or the handheld apparatus. The instructions can include a variety of items including instructing the presentation of a textual representation of dictation, or a function or command to be executed by the handheld device (such as linking to a website, opening a file, cutting, pasting, saving, or other file menu type functionalities), or by the computing device itself.

Abstract: A method and apparatus that manages speech decoders in a communication device may include detecting a change in transmission rate from a higher rate to a lower rate, decoding and shifting a first, second and third received first decoder set of frame parameters, generating a first decoder output audio frame from the previously shifted frame parameters, generating a first, second and third second decoder audio fill frame, the second decoder being a higher rate decoder than first decoder, outputting a first and second second decoder audio fill frame, combining the first decoder audio frame and the third second decoder audio fill frame with overlapping triangular windows, and outputting combined first decoder and second decoder frames to an audio buffer for subsequent transmission to a user of the communication device. In an alternative embodiment, another method may include detecting and processing a change in transmission rate from a lower rate to a higher rate.

Abstract: A data reproduction device is provided for achieving seamless reproduction of a stream where a validity of a bandwidth extension function is switched in the stream. The data reproduction device includes an input frequency obtainment unit analyzing header information Hdr and obtaining an input frequency FSin, which is the frequency of basic data, an output frequency determination unit performing predetermined processing based on the input frequency FSin and determining an output frequency FSout, which is the sampling frequency of a decoded frame Fdata, and a decoding unit (2003) which, if the SBR function is valid in a frame to be decoded, decodes sample data at the input frequency FSin and extends the bandwidth of the sampling frequency up to the output frequency FSout, while if the SBR function is not valid in the frame, upsamples the decoding result obtained at the input frequency FSin to the output frequency FSout.

Abstract: Systems and methods for updating electronic calendar information. Speech is received from a user at a vehicle telematics unit (VTU), wherein the speech is representative of information related to a particular vehicle trip. The received speech is recorded in the VTU as a voice memo, and data associated with the voice memo is communicated from the VTU to a computer running a calendaring application. The data is associated with a field of the calendaring application, and stored in association with the calendaring application field.

Abstract: In one embodiment, the method includes receiving an audio data frame having at least first and second channels. The first and second channels are independently subdivided into blocks if the first and second channels are not correlated with each other. The first and second channels are decoded, and the subdivided blocks of the first and second channels are not interleaved if the first and second channels are independently subdivided.

Abstract: An apparatus for automatic dissection of segmented audio signals, wherein at least one information signal for identifying programs included in said audio signals and for identifying contents included in said programs. Content detection device detects programs and contents belonging to the respective programs in the information signal. Program weighting device weights each program includes in the information signal based on the contents of the respective program detected by the content detection device. Program ranking device indentifies programmers of the same category and ranking said programs based on a weighting result for each program provided by the program weighting device.

Abstract: A system and method to capture and analyze image data of an object, the system including: an end unit to capture the image data, the end unit including: an image capturing unit to capture the image data of the object, and a transmitting unit to transmit the captured image data; and a data analysis server to analyze the image data, the data analysis server including: a receiving unit to receive the transmitted image data, and a control unit to analyze the received image data by performing one or more machine vision functions on the received image data, and to control a transmitting of a result of the one or more machine vision functions to the end unit and/or a performing of a function according to the result of the one or more machine vision functions. Accordingly, complex image data of an object can be collected by a remote image capturing device in order to identify the object by analyzing the image data in a separate server.

Abstract: First encoded voice bits are transcoded into second encoded voice bits by dividing the first encoded voice bits into one or more received frames, with each received frame containing multiple ones of the first encoded voice bits. First parameter bits for at least one of the received frames are generated by applying error control decoding to one or more of the encoded voice bits contained in the received frame, speech parameters are computed from the first parameter bits, and the speech parameters are quantized to produce second parameter bits. Finally, a transmission frame is formed by applying error control encoding to one or more of the second parameter bits, and the transmission frame is included in the second encoded voice bits.

Abstract: In one example, a mobile device encodes a digital bitstream using a particular set of modulation parameters to generate an audio signal that has different audio tones selected to pass through a vocoder of the mobile device. The particular set of modulation parameters is optimized for a subset of a plurality of vocoding modes without a priori knowledge of which one of the vocoding modes is currently operated by the vocoder. The mobile device conducts transmissions over the wireless telecommunications network through the vocoder using the particular set of modulation parameters, and monitors these transmissions for errors. If the errors reach a threshold, then the vocoder may be using one of the vocoding modes that are not included in the subset for which the particular set of modulation parameters is optimized, and accordingly, the modulation device switches from the particular set of modulation parameters to a different set of modulation parameters.

Abstract: Provided is an apparatus for integrally encoding and decoding a speech signal and an audio signal. An encoding apparatus for integrally encoding a speech signal and an audio signal, may include: a module selection unit to analyze a characteristic of an input signal and to select a first encoding module for encoding a first frame of the input signal; a speech encoding unit to encode the input signal according to a selection of the module selection unit and to generate a speech bitstream; an audio encoding unit to encode the input signal according to the selection of the module selection unit and to generate an audio bitstream; and a bitstream generation unit to generate an output bitstream from the speech encoding unit or the audio encoding unit according to the selection of the module selection unit.

Abstract: A system for leaving and transmitting speech messages automatically analyzes input speech of at least a reminder, fetches a plurality of tag informations, and transmits speech message to at least a message receiver, according to the transmit criterions of the reminder. A command or message parser parses the tag informations at least including at least a reminder ID, at least a transmitted command and at least a speech message. The tag informations are sent to a message composer for being synthesized into a transmitted message. A transmitting controller controls a device switch according to the reminder ID and the transmitted command, to allow the transmitted message send to the message receiver via a transmitting device.

Abstract: A multi-stream voice transmission system includes a transmitting terminal and a receiving terminal for transmitting and receiving first and second packet streams via first and second network channels. The receiving terminal includes a playout buffer for buffering the first and second packet streams, generates an output voice signal from the buffered packets according to a playout schedule adjusting coefficient ?, generates packet loss parameters and packet delay parameters corresponding to loss and delay experienced by the first and second packet streams, and provides the parameters to the transmitting terminal.

Abstract: A sound coding device having a monaural/stereo scalable structure and capable of efficiently coding stereo sound. even when the correlation between the channel signals of a stereo signal is small. In a core layer coding block of this device, a monaural signal generating section generates a monaural signal from first and second-channel sound signal, a monaural signal coding section codes the monaural signal, and a monaural signal decoding section greatest a monaural decoded signal from monaural signal coded data and outputs it to an expansion layer coding block. In the expansion layer coding block, a first-channel prediction signal synthesizing section synthesizes a first-channel prediction signal from the monaural decoded signal and a first-channel prediction filter digitizing parameter and a second-channel prediction signal synthesizing section synthesizes a second-channel prediction signal from the monaural decoded signal and second-channel prediction filter digitizing parameter.

Abstract: Generic and specific C-to-E binaural cue coding (BCC) schemes are described, including those in which one or more of the input channels are transmitted as unmodified channels that are not downmixed at the BCC encoder and not upmixed at the BCC decoder. The specific BCC schemes described include 5-to-2, 6-to-5, 7-to-5, 6.1-to-5.1, 7.1-to-5.1, and 6.2-to-5.1, where “0.1” indicates a single low-frequency effects (LFE) channel and “0.2” indicates two LFE channels.

Abstract: Provided are, among other things, systems, methods and techniques for decoding an audio signal from a frame-based bit stream. Each frame includes processing information pertaining to the frame and entropy-encoded quantization indexes representing audio data within the frame. The processing information includes: (i) code book indexes, (ii) code book application information specifying ranges of entropy-encoded quantization indexes to which the code books are to be applied, and (iii) window information. The entropy-encoded quantization indexes are decoded by applying the identified code books to the corresponding ranges of entropy-encoded quantization indexes. Subband samples are then generated by dequantizing the decoded quantization indexes, and a sequence of different window functions that were applied within a single frame of the audio data is identified based on the window information.

Abstract: An audio signal is encoded by a first waveform encoder (103) to generate a first waveform based bit-stream component. A second encoder (105) encodes the audio signal to generate a second bit-stream component comprising first enhancement data and a third encoder (107) encodes the audio signal to generate a third bit-stream component comprising second enhancement data for the first waveform based bit-stream component. The first and second bit-stream components correspond to a first representation of the audio signal and the first and third bit-stream components correspond to a second representation of the audio signal. A scalable audio bit-stream is generated by a bit-stream generator (109). The different representations may be selected between by a decoder thereby allowing a flexible and scalable bit-stream to be communicated. The second encoder (105) may specifically be a waveform encoder and the third encoder (107) may specifically be a parametric encoder.

Abstract: A communications network is used to transfer user attribute information about participants in a communication session to their respective communication terminals for storage and use thereon to configure a speech codec to operate in a speaker-dependent manner, thereby improving speech coding efficiency. In a network-assisted model, the user attribute information is stored on the communications network and selectively transmitted to the communication terminals while in a peer-assisted model, the user attribute information is derived by and transferred between communication terminals.

Abstract: A serial transmission system has a transmission signal generator and a transmission signal receiver. The transmission signal generator generates a digital audio signal of multi-channel based on information of a sampling frequency, and serially transmits the information of the sampling frequency together with the digital audio signal. The transmission signal receiver serially receives the information of the sampling frequency and the digital audio signal from the transmission signal generator, and detects change in a transmission clock based on the information of the sampling frequency.

Abstract: A signal classifying method and apparatus are disclosed. The signal classifying method includes: obtaining a spectrum fluctuation parameter of a current signal frame determined as a foreground frame, and buffering the spectrum fluctuation parameter; obtaining a spectrum fluctuation variance of the current signal frame according to spectrum fluctuation parameters of all buffered signal frames, and buffering the spectrum fluctuation variance; and calculating a ratio of signal frames whose spectrum fluctuation variance is above or equal to a first threshold to all the buffered signal frames, and determining the current signal frame as a speech frame if the ratio is above or equal to a second threshold or determining the current signal frame as a music frame if the ratio is below the second threshold.

Abstract: System and method for encoding, transmitting and decoding audio data. Audio bit steam syntax is re-organized to allow system optimizations that work well with memory latency and memory burst operations. Multiple small entropy coding tables are stored in RAM and loaded to on-chip memory as needed. Audio prediction is pipelined in the bitstream syntax. Intra frames, independent of other frames in the bitstream, are included in the bitstream for error recovery and channel change. New algorithms are implemented in legacy syntax by including the new information in the user data space of the audio frame. The new decoder can use projection to determine where the new information is and read ahead in the stream. Audio prediction from the immediately previous frame is restricted. Audio prediction is performed across channels within a single audio frame. A variable re-order function comprises storing channels of data to DRAM in the order they are decoded and reading them out in presentation order.

Abstract: A method and apparatus for generating a control signal for processing a speech signal comprising the steps of: adjusting the signal relative to a threshold level; and responsive to detection of a falling edge of the signal, holding the signal level for a holding period. The technique further comprises ‘slowing’ each rising edge of the signal. The technique further comprises attenuating each falling edge of the signal. The steps are carried out on a signal representing the envelope of the speech signal.

Abstract: Monitoring accuracy degrades due to a noise in an environment where there are many sound sources other than those to be monitored. Easy initialization is required for an environment where many apparatuses operate. A sound monitoring system includes a microphone array having multiple microphones and a location-based abnormal sound monitoring section as a processing section. The location-based abnormal sound monitoring section is supplied with an input signal from the microphone array via a waveform acquisition section and a network. Using the input signal, the location-based abnormal sound monitoring section detects a temporal change in a sound source direction histogram. Based on a detected change result, the location-based abnormal sound monitoring section checks for abnormality in a sound field and outputs a monitoring result. The processing section searches for a microphone array near the sound source to be monitored.

Abstract: Embodiments of the invention include apparatuses, systems, computer readable media, and methods for processing speech signals in a manner that enhances capacity, efficiency and hardware utilization of a communications network. A method, according to one embodiment, includes receiving speech signals, determining a subchannel power imbalance ratio of at least two subchannels, and selecting a receiver architecture for processing the speech signals in accordance with the determined subchannel power imbalance ratio.

Abstract: A method and device for updating statuses of synthesis filters are provided. The method includes: exciting a synthesis filter corresponding to a first encoding rate by using an excitation signal of the first encoding rate, outputting reconstructed signal information, and updating status information of the synthesis filter and a synthesis filter corresponding to a second encoding rate. In the present disclosure, the status of the synthesis filter corresponding to the current rate and the statuses of the synthesis filters at other rates are updated. Thus, synchronization between the statuses of the synthesis filters corresponding to different rates at the encoding terminal may be realized, thereby facilitating the consistency of the reconstructed signals of the encoding and decoding terminals when the encoding rate is switched, and improving the quality of the reconstructed signal of the decoding terminal.

Abstract: The invention relates to an audio encoder and decoder and methods for audio encoding and decoding. In a preferred encoder embodiment an audio signal is encoded by deterministic encoder means to form a first encoded signal part. A spectrum of the audio signal is determined and represented by an excitation pattern, i.e. spectral values corresponding to human auditory filters, as a second encoded signal part. A masking curve is also extracted based on the excitation pattern, thus improving encoding efficiency in terms of bit rate. In a preferred decoder the first encoded signal part is decoded by deterministic decoder means. A noise generator uses the decoded first signal part together with the second signal part, i.e. the excitation pattern for the original audio signal, to generate a noise signal. The noise signal is then added to the first decoded signal part to form an output audio signal. At the decoder side the masking curve is also extracted based on the second encoded signal part, i.e.

Abstract: A reproduction apparatus that reproduces compressed audio data recorded in a recording medium inserts dummy data between data to be concatenated and reproduces the data when performing a specific reproduction of the data obtained by concatenating data which are discontinuously read from the recording medium.

Abstract: A method of creating and storing a marked location in a portable electronic device includes receiving current position information of the portable electronic device, recording description data for a marked location when a user of the portable electronic device is located at the marked location, and storing the description data and position information corresponding to the marked location in a memory.

Abstract: A sending apparatus includes a compressed music data outputting unit that outputs one of compressed music data including compressed basic music data in a first mode and compressed music data including the basic music data as well as high frequency information for extending high frequency of the basic music data in a second mode, and a formatter that transmits a compressed music data outputted by the compressed music data outputting unit first and then music information including a value indicating which mode is used for the compressed music data.

Abstract: A method for determining a bit boundary of a repetition-coded signal including bits each having a plurality of epochs includes (a) counting the epochs repeatedly from an initial number to a predetermined number in a predetermined time, (b) sensing sign changes in the epochs, (c) recording each sensed sign change with a weighting function to a corresponding counting number of the epoch, and (d) determining the bit boundary according to a result of step (c).

Abstract: Methods, systems and devices for creating special communications or recordings containing messages from family members and the like may be delivered to a recipient in the form of a keepsake type of device that can be used to play the communications when and as many times as desired.

Abstract: Embodiments of the invention provides a method and device for assigning bitrates to a plurality of channels in a scalable audio encoding/truncation process. Different bitrates are assigned to different channels in the scalable audio encoding/truncation process.

Abstract: One embodiment of the present invention provides a method of determining an evidence value capturing whether two band-pass signals are harmonics of a common fundamental frequency. A further embodiment of the present invention evaluates the distance between significant points of a signal such as a sinusoidal signal. One embodiment of the present invention provides a method of determining whether two or more band-pass signals are harmonics of a fundamental frequency, comprising evaluating a first distance between a first set of two or more significant points of a first band-pass signal, evaluating a second distance between a second set of two or more significant points of a second band-pass signal, and comparing the first distance to the second distance to determine whether the first and second signals are harmonics of the fundamental frequency.

Abstract: The invention concerns a method and apparatus for performing packet loss or Frame Erasure Concealment (FEC) for a speech coder that does not have a built-in or standard FEC process. A receiver with a decoder receives encoded frames of compressed speech information transmitted from an encoder. A lost frame detector at the receiver determines if an encoded frame has been lost or corrupted in transmission, or erased. If the encoded frame is not erased, the encoded frame is decoded by a decoder and a temporary memory is updated with the decoder's output. A predetermined delay period is applied and the audio frame is then output. If the lost frame detector determines that the encoded frame is erased, a FEC module applies a frame concealment process to the signal. The FEC processing produces natural sounding synthetic speech for the erased frames.

Abstract: A method of controlling an output of an ultrasonic speaker that reproduces audible-frequency-band signal sounds by modulating carrier waves with audible-frequency-band signal waves output from a signal source and driving an ultrasonic transducer with the modulated waves includes: dividing the audible-frequency-band signal waves into a plurality of frequency bands; separately adjusting amplitudes of the signal waves and amplitudes of the modulated waves in the respective frequency bands; and driving a plurality of ultrasonic transducers provided corresponding to the respective frequency bands with the modulated waves generated corresponding to the respective frequency bands.