Abstract: An audio player includes a storage unit storing music type modes and audio files; an input device provided for selecting an automatic switching mode of the audio player; a detection circuit sampling noise of the current environment and generating a detection signal according to the noise of the current environment; a processor performing the automatic switching mode by selecting one of the music type modes according to the detection signal and finding one or more of the audio files having identities corresponding to the selected music type mode; and an output device outputting data of the audio files having identities corresponding to the selected music type mode.

Abstract: A digital multi-media device provides features for a user unskilled in musical arts or sound handling techniques that provides automatic musical score composition in accordance with contained composition instructions. Stored sound samples and interfaces for obtaining external signals provide signals for merger with visual and sound presentations to obtain altered presentations either time shifted or in real time. In this fashion the user can create simulated radio stations for playback of prearranged and composed audio material. Further, the automatically composed musical score may be mixed with synthesized, digitized signals from the stored sound samples and external signals obtained through the device interfaces.

Abstract: In a waveform compressing apparatus, a trial mode selecting portion selects a trial mode having the highest compression rate from a plurality of candidate modes which have not been selected before as a trial mode for generating a residue code, the selected trial mode comprising a scalar quantization mode or a vector quantization mode. A waveform data compressing portion compresses a given data amount of original waveform data according to the selected trial mode so as to generate the residue code, the data amount being determined in correspondence with the selected trial mode. A waveform data restoring portion generates a restored waveform data from the compressed data using the generated residue code. A determining portion measures an evaluation value of a quantization error contained in the restored waveform data relative to the original waveform data, and determines whether the evaluation value is equal to or smaller than a predetermined allowable value.

Abstract: A method and system for provides a user with an ability to capture a sample of an experiential environment and deliver that sample to an interactive service to trigger one or more predetermined events. In exemplary embodiments of the invention such triggered events include the delivery of information and services to the user, the execution of tasks and instructions by the service on the user's behalf, communication events; surveillance events and other control-oriented events that are responsive to the user's wishes. In other exemplary embodiments of the invention, the triggered events include transaction-oriented events, entertainment events, and events associated with enhancements to human ability or function.

Abstract: An improved arbitrary waveform generator has a waveform memory for storing digitized waveforms, a waveform playout for playing out desired ones of the digitized waveforms as analog waveforms and a sequencer for controlling the waveform playout, the sequencer providing indications of the desired waveform for playout and a desired starting sample position for the desired waveform. The sequencer includes a tracking mechanism for the desired waveform so that the desired waveform is phase coherent when playout is interrupted and restarted later according to programming of the sequencer.

Abstract: Described is a technology by which a playback list comprising similar songs is automatically built based on automatically detected/generated song attributes, such as by extracting numeric features of each song. The attributes may be downloaded from a remote connection, and/or may be locally generated on the playback device. To build a playlist, a seed song's attributes may be compared against attributes of other songs to determine which other songs are similar to the seed song and thus included in the playlist. Another way to build a playlist is based on similarity of songs to a set of user provided-attributes, such as corresponding to moods or usage modes such as “resting” “reading” “jogging” or “driving” moods/modes. The playlist may be dynamically adjusted based on user interaction with the device, such as when a user skips a song, queues a song, or dequeues a song.

Abstract: A file creation process, file format and playback device are provided that enables an interactive and if desired collaborative music playback experience for the user(s) by combining or retrofitting an “original song” with a MIDI time grid, the MIDI score of the song and other data in a synchronized fashion. The invention enables a music interaction platform that requires a small amount of time to learn and very little skill, knowledge or talent to use and is designed to bring “mixing music” to the average person.

Abstract: Methods, computing devices, and machine readable storage media for generating a fingerprint of a music sample. The music sample may be filtered into a plurality of frequency bands. Onsets in each of the frequency bands may be independently detected. Inter-onset intervals between pairs of onsets within the same frequency band may be determined. At least one code associated with each onset may be generated, each code comprising a frequency band identifier identifying a frequency band in which the associated onset occurred and one or more inter-onset intervals. Each code may be associated with a timestamp indicating when the associated onset occurred within the music sample. All generated codes and the associated timestamps may be combined to form the fingerprint.

Abstract: A method and computer based program which performs a series of steps for automatically and accurately determining each note played in a song for each instrument and vocal. The method and program can transcribe or create sheet music for each individual instrument, as well as provide the ability to remove any combination of or individual instruments or vocal track basically from nearly any existing song, or future songs.

Abstract: A music sound generation apparatus generates music-sound data based on sound source data, and outputs the music-sound data in synchronization with audio data obtained separately from the music-sound data. A channel assigner assigns a predetermined number of channels to the audio data and the music-sound data. A read control unit reads the audio data and the music-sound data in accordance with their assigned channels. A signal processing unit generates predetermined music by controlling the sound level of the read audio data and the read music-sound data on each assigned channel and mixing the audio data and the music-sound data. When detecting no unoccupied channel, the channel assigner determines, as an unoccupied channel, a channel selected from channels excluding the channel to which the audio data has been assigned. The signal processing unit performs sound elimination on the unoccupied channel.

Abstract: A synthetic multi-string musical instrument captures a stream of expressive gestures indicated on a multi-touch sensitive display for note/chord soundings and associated performance effects and embellishments. Visual cues in accord with a musical score may be revealed/advanced at a current performance tempo, but it is the user's gestures that actually drive the audible performance rendering via digital synthesis. Opportunities for user expression (or variance from score) include onset and duration of note soundings, tempo changes, as well as uncued string bend effects, vibrato, etc. Gesturing mechanism are provide to allow user musicians to sound chords without having to register precisely accurate multi-touch screen contacts. This can be especially helpful for mobile phone, media player and game controller embodiments, where there is generally limited real-estate to display six (6) or more strings, and user fingers are generally too fat to precisely contact such strings.

Abstract: A musical performance apparatus has a waveform memory WM in which sample values indicative of waveforms of a plurality of tones are stored so that the sampling periods correspond to addresses. The musical performance apparatus also has a tone generation circuit 15 which can repeatedly reproduce a section of the tone. A loop top address and loop end address corresponding to the top and end of the section of the first tone, respectively, are designated. When a reading address for reading the sample values of the first tone has reached a certain address, the tone generation circuit 15 changes the designated loop top address and the loop end address to addresses corresponding to the top and end of the section of the second tone so that the reproduction of the section of the second tone will start at a position which is situated in the section of the second tone and corresponds to the certain address.

Abstract: One aspect provides a method, including: receiving an analog note unit at an information handling device; storing note unit information corresponding to the analog note unit in a memory device of the information handling device; identifying the analog note unit input using the note unit information; and providing a result output to a user responsive to identifying the analog note unit. Other aspects are described and claimed.

Abstract: Melody and accompaniment audio signals are received and processed to identify one or more harmony notes and a harmony signal is produced based on the one or more harmony notes. Typically the melody note is identified and a spectrum of the accompaniment audio signal and is obtained, and one or more harmony notes are identified based on the melody note and the accompaniment spectrum. The melody, and accompaniment signals can be processed in real-time for combination with the harmony signal in an audio performance. In some examples, audio signals are processed and harmonies generated for subsequent performance based on, for example, MIDI files generated from the audio signals.

Abstract: Methods and apparatus for relative pitch tracking of multiple arbitrary sounds. A probabilistic method for pitch tracking may be implemented as or in a pitch tracking module. A constant-Q transform of an input signal may be decomposed to estimate one or more kernel distributions and one or more impulse distributions. Each kernel distribution represents a spectrum of a particular source, and each impulse distribution represents a relative pitch track for a particular source. The decomposition of the constant-Q transform may be performed according to shift-invariant probabilistic latent component analysis, and may include applying an expectation maximization algorithm to estimate the kernel distributions and the impulse distributions. When decomposing, a prior, e.g. a sliding-Gaussian Dirichlet prior or an entropic prior, and/or a temporal continuity constraint may be imposed on each impulse distribution.

Abstract: A flash-memory based stored-sample electronic music synthesizer enables the electronic reproduction of a large number of independent voices while accommodating the exacting demands of voice continuity, minimal note-start latency, and voice synchronicity. Error correction code associated with each page of a sound sample is stored with the sound sample and is retrieved with the sound sample during playback to thereby increase the overall sample retrieval rate.

Abstract: A flash-memory based stored-sample electronic music synthesizer enables the electronic reproduction of a large number of independent voices while accommodating the exacting demands of voice continuity, minimal note-start latency, and voice synchronicity.

Abstract: A double talk detector for controlling the echo path estimation in a telecommunication system by indicating when a received coded speech signal is dominated by a non-echo signal; i.e., that so-called double talk exists. This is determined by extracting LSPs from a coded speech frame of the received coded speech signal when the signal power exceeds a first threshold value, converting each of said extracted LSPs into LSFs, and calculating the distance between each two adjacent LSFs. For each distance that is smaller than a second threshold, a spectral peak is located between the two LSFs, and it is determined whether said spectral peak is an echo or not. When a predetermined number of non-echo spectral peaks are located in the received speech signal, double talk will be indicated, and the echo path estimation may be disabled.

Abstract: A musical tone signal generating apparatus has waveform memory WM which stores a plurality of compressed data sets obtained by compressing, by linear prediction, sample values obtained by sampling musical tones. The musical tone signal generating apparatus has cache circuit 740 which reads out compressed data from waveform memory WM within an assigned computing period in response to instructions to generate a musical tone, and decoding circuit 750 which decodes the compressed data and outputs the decoded data as the data indicative of a sample value. The musical tone signal generating apparatus has CPU 901 inputs tone pitch information indicative of a tone pitch of a musical tone which is to be generated, identifies waveform data which is to be read out by cache circuit 740 from waveform memory WM and determines, in accordance with the identified waveform data, the length of the computing period which is to be assigned.

Abstract: This disclosure relates to a method for recording and compiling a music section, wherein multiple takes of the music section are recorded; and wherein the method uses a take folder to store the multiple takes. The take folder may belong to a single track. Sections of one or more takes can be selected to be part of a resulting compilation which is composed of the selected sections. In one embodiment, a method automatically ensures that selected sections do not overlap in time and that there are no breaks between adjacent selections. In addition, the present invention relates to programs, a storage media and data processing systems having the above features as well as to a storage medium containing the music produced thereby.

Abstract: A sound effect sensing turntable for a digital music broadcaster includes a capacitive charge sensing plate able to independently detect the variation of charge. A signal transmission unit is disposed between the sensing plate and a control plate for transmitting signals and thus, on receiving a signal, the control plate will produce relative sound effect to be output. In operating, when a DJ's fingers slightly touch the turntable, the sensing plate will sense charge variation on the turntable and actuate the control plate to control corresponding sound effect. By detecting the variation of charge on the turntable for controlling sound effect, the mode of touching the turntable of this invention for producing sound effect is like that of conventionally controlling a vinyl record for controlling sound effect, getting rid of conventional defect of producing wrong sound effect due to assembly errors.

Abstract: Methods for generating a new recording of a past musical performance of a musician from a recording of the past musical performance include obtaining a high-resolution data record representing actions of the musician while playing the past musical performance that is generated based on the recording of the past musical performance and positioning an automated musical instrument in a selected acoustic context and a sound detection device at a selected sound detection location in the selected acoustic context. The high-resolution data record is provided to the musical instrument to cause the musical instrument to re-produce the actions of the musician while playing the past performance. Sound waves generated by the musical instrument are recorded while the actions of the musician are being re-produced to generate the new recording of the past musical performance.

Abstract: The sound-processing apparatus of the present invention generates plural frequency data by decoding plural encoded sound data and applying inverse quantization. Each of the frequency data are subjected to sound-processing and then synthesized into one single frequency data. Transformation processing from frequency domain to time domain is applied to the synthesized single frequency data so as to generate sound data in time domain so as to reduce computation amounts of decoding process.

Abstract: A music piece reproducing apparatus has: a sound collecting unit collecting noise data of an ambient environment; an analyzing unit performing a frequency analysis to the noise data collected by the sound collecting unit and extracting a feature amount showing a feature of the noise data; a noise parameter database classifying the noise data every environment; and a reproducing unit reproducing music piece data which has been recorded in a recording medium and classified every category, wherein the analyzing unit classifies the noise data with reference to the noise parameter database on the basis of the extracted feature amount of the noise data, selects the previously-corresponded category of music pieces in accordance with a classification result of the noise data, and selects predetermined music piece data from music piece data classified into the selected category.

Abstract: Systems for providing a tune master caller identification (ID) device include a telecommunications network and a caller device configured to provide caller identification information for an incoming telephone call, and a tune master caller ID device that is coupled to a telephone device. The tune master caller ID device receives caller identification information, associates a tune with the caller identification information, and plays the tune associated with the caller identification. The telephone device coupled to the tune master caller ID device provides telephone call processing capabilities. Other systems and methods are also provided.

Abstract: A system and associated methods provide an audio experience such that a user spatially perceives one or more audio events. One particular method set forth involves obtaining audio events and presenting the audio events so that an audio experience is provided. According to one embodiment of the method, upon obtaining audio events, the audio events are associated with one or more corresponding audio components. Thereafter, the audio experience is determined based on the audio events and associated audio components. The audio experience is then presented such that the user may spatially perceive the audio events.

Abstract: The apparatus for converting an information signal from a time to a variable spectral representation includes a means for windowing the information signal, a means for converting the windowed information signal to a spectral representation, and a means for weighting a set of information signal spectral coefficients with several sets of complex base function coefficients provided from a means for providing the sets of base function coefficients. The sets of base function coefficients are derived from base functions of various frequencies by windowing and transform, wherein several sets of base function coefficients are provided for one and the same base function for base functions of higher frequencies, wherein the windows for providing these sets are related to various time portions of the base function. The variable spectral representation exhibits variable bandwidth of the variable spectral coefficients, which are efficient and accurate to calculate and especially suited for music analysis purposes.

Abstract: Methods and systems for providing lyric information for a song within an audio signal. This may be done, for example, to allow a user to sing along with a song the user hears on a radio. To provide the lyric information, an interactive service may be accessed. A sample of an audio signal that includes at least a portion of the song may be captured. The sample of the audio signal may provided to the interactive service. Lyric information may then be received for the song at a user device. The user device may also display the lyric information in synchrony with the rendering of the song within the audio signal to, for example, allow the user to sing along with the song.

Abstract: Provided is a waveform generating apparatus that generates a signal having an arbitrary waveform, comprising a waveform memory that stores a plurality of pieces of waveform data that each include a sequence of signal values; a filtering section that (i) reads from the waveform memory a piece of waveform data serving as a basis for a waveform to be generated, from among the plurality of pieces of waveform data, (ii) performs a conversion by filtering the read piece of waveform data to obtain a piece of converted waveform data, and (iii) writes to the waveform memory the piece of converted waveform data; and a waveform output section that reads the piece of converted waveform data from the waveform memory and outputs a signal having a waveform corresponding to the sequence of signal values of the read piece of converted waveform data.

Abstract: Characteristic amounts are calculated from each of multiple musical piece contents, and then time points when the characteristic amounts exceed a threshold are detected as cut points. Information on the cut point concerning each of the multiple musical piece contents is stored. Based on the stored information on the cut point, a judgment is made as to whether the multiple musical piece contents match each other.

Abstract: A device may retrieve a ring tone, an image, and a visual effect from a database based on a client line identifier. In addition, the device may generate frequency samples based on the ring tone, may transform the image based on the frequency samples and the visual effect, and may display the transformed image.

Abstract: The disclosed subject matter includes, in one aspect, a signal processor having circuitry configured to substitute an assortment of pre-recorded sounds in place of the sound of the guitar. The pre-recorded sounds are triggered by a guitar note that is above a certain threshold. The threshold can be varied, as can the length of time of the substitution.

Abstract: There is provided a signal processing device for processing an audio signal, the signal processing device including: an onset time detection unit for detecting an onset time based on a level of the audio signal; and a beat length calculation unit for obtaining a beat length Q by: setting an objective function P(Q|X) and an auxiliary function, the objective function P(Q|X) representing a probability that, when an interval X between the onset times is given, the interval X is the beat length Q, the auxiliary function being for inducing an update of both the beat length Q and a tempo Z that results in a monotonous increase of the objective function P(Q|X); and repeating maximization of the auxiliary function to have the auxiliary function converge.

Abstract: A digital multi-media device provides features for a user unskilled in musical arts or sound handling techniques that provides automatic musical score composition in accordance with contained composition instructions. Stored sound samples and interfaces for obtaining external signals provide signals for merger with visual and sound presentations to obtain altered presentations either time shifted or in real time. In this fashion the user can create simulated radio stations for playback of prearranged and composed audio material. Further, the automatically composed musical score may be mixed with synthesized, digitized signals from the stored sound samples and external signals obtained through the device interfaces.

Abstract: A music modification system is provided and described herein, which includes: a) a computer, b) a music element library, c) at least one part-score database, d) a software code that executes a music modification system on the computer, wherein the music modification system accesses or manipulates the information in the music element library and accesses the at least one part-score database, and e) a graphical or audio user interface that is coupled to the computer. Methods of modifying a musical score or piece are described herein and include: a) providing a music element library, b) providing at least one part-score database, wherein the database comprises at least one music score, at least one music pieces, at least one music part or a combination thereof, c) providing an executable music modification system, and d) utilizing the music modification system and the music element library to modify at least part of the at least one part-score database.

Abstract: This disclosure describes techniques that make use of a waveform fetch unit that operates to retrieve waveform samples on behalf of each of a plurality of hardware processing elements that operate simultaneously to service various audio synthesis parameters generated from one or more audio files, such as musical instrument digital interface (MIDI) files. In one example, a method comprises receiving a request for a waveform sample from an audio processing element, and servicing the request by calculating a waveform sample number for the requested waveform sample based on a phase increment contained in the request and an audio synthesis parameter control word associated with the requested waveform sample, retrieving the waveform sample from a local cache using the waveform sample number, and sending the retrieved waveform sample to the requesting audio processing element.

Abstract: In general, the techniques of this disclosure may be used to control utilization of bandwidth allocated to an audio processing module. For example, to process various audio synthesis parameters, the audio processing module may retrieve reference waveform samples for use in generating audio information for voices within an audio frame, such as a MIDI frame. In some cases, the amount of bandwidth available for retrieving the reference waveforms from memory is limited. To manage the utilization of the allocated bandwidth a bandwidth control module estimates an amount of bandwidth required to retrieve reference waveforms for all the voices of the audio frame, and selects one or more voices to be eliminated from generated audio information when the bandwidth estimate exceeds the allocated bandwidth.

Abstract: There are disclosed methods and apparatus for understanding music. A classifier machine may be trained for each of a plurality of selected terms using a first plurality of music samples. The classifier machines may then be tested using a second plurality of music samples. The results from testing the classifier machines may then be used to select a plurality of semantic basis function from the selected terms. A semantic basis classifier machine may then be trained for each semantic basis function.

Abstract: A digital sound effect processing device includes an analog/digital converter, a digital signal processor, and a digital/analog converter. The analog/digital converter converts analog frequency signals input by a signal input unit into digital signals to be output. The digital signal processor has one input terminal connected with the analog/digital converter and another input terminal connected with a music access unit for collecting digital audio data input by the analog/digital converter and original music data stored in the music access unit and changing the broadcasting sound effects of the original music data according to the digital audio data and then digitally outputting the sound effects. The digital/analog converter converts the digital music output by the digital signal processor into analog music to be broadcast through a loudspeaker.

Abstract: A flash-memory based stored-sample electronic music synthesizer enables the electronic reproduction of a large number of independent voices while accommodating the exacting demands of voice continuity, minimal note-start latency, and voice synchronicity.

Abstract: In a musical tone generating apparatus 100, sampling waveform data sampled by a sampling inputting section 15 is, after being divided into an n-number of segments, stored in divided areas W(1) to W(n) of a sampling waveform data area SWDE, respectively. When a chord is played in which a subsequent key is pressed within a predetermined time after the preceding key is pressed (Step SC4 to Step SC5), sampling sounds are produced at the respective pitches of the pressed keys with the use of sampling waveform data in the same divided area W(n) designated by the same pointer n value as that when the preceding key is pressed (Step SC6).

Abstract: This disclosure describes techniques that make use of a summing buffer that receives waveform samples from audio processing elements, and sums and stores the waveform sums for a given frame. In one example, a method comprises summing a waveform sample received from an audio processing element to produce a waveform sum associated with a first audio frame, storing the waveform sum in a memory, wherein the memory is logically partitioned into a plurality of memory blocks, and locking memory blocks containing the waveform sum associated with the first audio frame, transferring contents of locked memory blocks to an external processor, unlocking a memory block after contents of the memory block have been transferred to the external processor, and storing a waveform sum associated with a second audio frame within the unlocked memory block concurrently with transferring contents of remaining locked memory blocks associated with the first audio frame.

Abstract: An alert signal of a communication device (1), such as the ring signal of a mobile phone, comprises a set of musical samples that are played by the communication device in sequence. The sequence may be randomly generated, either in response to the event causing the alert or in a set up phase. Alternatively, the user of the device can create a tune to be used as the alert signal by creating a sequence of samples. The user selects samples from a list of samples to create the sequence of samples as individual samples may appear more than once in the tune.

Abstract: Audio samples corresponding to audio extracts or whole audio titles are automatically mapped to triggers 12 in a playable sound-producing device 1, the mapping being dependent on the meta-data associated with the audio samples. Thus, a user can play the sound-producing device and generate sounds derived from his favorite audio titles. It is possible to define different mappings between the audio samples and the playable domain of the sound-producing device: an audio sample selector 50 can select between different possible samples for playback by comparing the audio properties of the samples and the play-mode and/or characteristics of the user's performance. An audio sampler/sample-processor 70 can automatically extract segments of an audio source file and map them to triggers in the sound-producing device 1.

Abstract: Herein described are at least a method and a system for assisting a musician in playing an instrument by way of using an intelligent multifunctional digital music display device (MDMDD). The multifunctional digital music display device (MDMDD) may be used to assist musicians or performers during practice sessions or performances. The various aspects of the invention provide a method of constructing an ADSR envelope for a note generated by an instrument, displaying one or more pages of music, generating a more accurate metronome beat consistent with the music being played, and alerting the musician when his instrument is out-of-tune. Further, the various aspects of the invention provide a method of statistically monitoring and reporting the performance of the musician. The system comprises a storage device capable of storing an instructional code, a processor for executing the instructional code, a microphone, and a display.

Abstract: The present disclosure relates to speech therapy and voice training methods. A system is provided which utilizes tonal and rhythm visualization components to allow a person to “see” their words as they attempt to improve or regain their ability to speak clearly. The system is also applicable to vocal music instruction and allows students improve their singing abilities by responding to visual feedback which incorporates both color and shape. The system may comprise a step-by-step instruction method, live performance abilities, and recording and playback features. Certain embodiments incorporate statistical analysis of student progress, remote access for teacher consultation, and video games for enhancing student interest.

Abstract: This disclosure relates to a method for recording and compiling a music section, wherein multiple takes of the music section are recorded; and wherein the method uses a take folder to store the multiple takes. The take folder may belong to a single track. Sections of one or more takes can be selected to be part of a resulting compilation which is composed of the selected sections. In one embodiment, a method automatically ensures that selected sections do not overlap in time and that there are no breaks between adjacent selections. In addition, the present invention relates to programs, a storage media and data processing systems having the above features as well as to a storage medium containing the music produced thereby.

Abstract: A method of generating a tone waveform using a CPU is provided which prevents or minimizes operational delays of other software processing that is executed concurrently with tone waveform generating processing on a multitask basis. The CPU collectively calculates 128 (one block of) tone waveform sample values each corresponding to a sampling clock pulse, and transmits the calculated tone waveform sample values to a reproduction section in response to a predetermined calculation triggering clock pulse generated every 128 samples. When sufficient processing capability of the CPU performing the multitask is available for the waveform sample calculation, tone waveform sample values for one or more following blocks are also calculated and stored in a sample buffer in advance. When the CPU is too busy with the other software processing to execute the waveform sample calculation, it is just sufficient that the previously stored tone waveform sample values be read out to be transmitted to the reproduction section.

Abstract: A sound source apparatus has operation blocks composed of softwares used to compute waveforms for generating a plurality of musical tones through a plurality of channels according to performance information. In the apparatus, a setting device sets an algorithm which determines a system composed of selective ones of the operation blocks systematically combined with each other to compute a waveform specific to one of the musical tones. A designating device responds to the performance information for designating one of the channels to be used for generating the musical tone. A generating device allocates the selective operation blocks to the one channel and systematically executes the allocated selective operation blocks according to the algorithm so as to compute the waveform to thereby generate the musical tone through the channel.

Abstract: A waveform data processing apparatus has a bus that transfers data signals representative of waveform data. A plurality of transmitting nodes transmit the data signals to the bus. A plurality of receiving nodes receive the data signals from the bus. A clock generator generates a word clock signal at each sampling period. A controller is responsive to the word clock signal for conducting a session of transferring the data signals within a sampling period, such that the transmitting nodes sequentially transmit the data signals in an order predetermined by the controller so as to avoid collision of the data signals within the sampling period, and each of the receiving nodes selectively admits a necessary one of the data signals outputted from the transmitting nodes and processes the admitted data signal within the sampling period.