Abstract: A method of constructing a waveform from N sampled data captured at N successive points in time, includes, in part, applying the N sampled data, K data at a time, to each of M delayed replicas of a filter that includes K taps so to generate N×M interpolated data. The waveform is then constructed from the N sampled data and the N×M interpolated data.

Abstract: A method for rendering lyrics is provided, including: acquiring pronunciation of a polyphonic word to be rendered in target lyrics, and acquiring playback time information of the pronunciation in the process of rendering the target lyrics; determining a first number of furiganas contained in the pronunciation; and word-by-word simultaneously rendering, according to the first number and the playback time information of the pronunciation of the polyphonic word to be rendered, the polyphonic word to be rendered and each furigana in the pronunciation of the polyphonic word to be rendered, wherein the pronunciation of the polyphonic word to be rendered is adjacent to and parallel to the polyphonic word to be rendered.

Abstract: Systems and methods for creating a digital audio filter, such as an impulse response filter, using a pickup audio signal provided to an instrument signal capture device and a microphone signal provided to a mobile device are disclosed. In one embodiment, a method includes capturing a first audio signal using a signal capture device, performing frequency analysis on the digitized first audio signal to generate a frequency response spectrum representation, transmitting the frequency response spectrum representation of the first audio signal from the signal capture device to a mobile device, capturing a second audio signal using a microphone on the mobile device, performing frequency analysis on the at least one digitized second audio signal using the mobile device to generate a frequency response spectrum representation, and generating a digital audio filter from the frequency response spectrum representations of the first audio signal and the second audio signal.

Abstract: Sound equipment includes: a beat position acquiring unit configured to analyze a beat position of music piece data to be reproduced or acquire an analysis result; and a beat position displaying unit configured to display a scale of a differential numerical value of the number of beats in a plus direction and a minus direction of the music piece data with reference to a currently reproduced position of the music piece data.

Abstract: An audio signal processing method, belongs to the field of terminal technologies. The audio signal processing method includes: acquiring a first audio signal of a target song sung by a user; extracting timbre information of the user from the first audio signal; acquiring intonation information of a standard audio signal of the target song; and generating a second audio signal of the target song based on the timbre information and the intonation information.

Abstract: An application for operating on a smart phone that records a musician's performance, either voice or instrumental, in combination with pre-recorded music. The combination allows for the auto tuning of the recording, the compression of the recording, the equalization of the recording, adding in reverb, correcting latency and the audio quantization of the rhythm, in addition to music enhancement features such as vocal spread, DeEsser, vocal doubler, vocal harmonizer, tape saturation, pitch correcdtion, flanger, phaser, auto pan, vibrato, tremolo, rotary, ring modulator, metalizer, expander, noise gate, wah, vocal leveling, tape stop, half speed, LoFi, and stutter. Once combined, the song is transmitted to social media and/or to an online store for sale. The user can also make a video with the song. Additional marketing such as song competitions or music reviews and ratings are also provided.

Abstract: An audio identification method and apparatus are disclosed within the technical field of audio processing technology. The audio identification solution includes obtaining an original pitch sequence of a to-be-identified audio, where the original pitch sequence is used to indicate a frequency of the to-be-identified audio at each time point. The audio identification solution further includes dividing the original pitch sequence into a plurality of pitch sub-sequences, respectively identifying the original pitch sequence and the plurality of pitch sub-sequences, and combining the identification results. In doing so, the audio identification solution obtains a final identification result by dividing a long pitch sequence into a plurality of short pitch sequences, thus respectively identifying the long pitch sequence and the plurality of short pitch sequences, and combining identification results.

Abstract: An audio processing device includes an acquisition unit that acquires audio that is picked up by a sound pick-up unit, a detector that detects an audio position of the audio, a determiner that determines whether or not the audio is a speech audio when the audio position is within a privacy protection area, an analyzer that analyzes the speech audio to acquire an emotion value, a converter that converts the speech audio into a substitute sound corresponding to the emotion value, and an output controller that causes an audio output that outputs the audio to output the substitute sound.

Abstract: A system and method eliminate the drag associated with CPU drag on experienced by combine audio streams from multiple sources. The embodiments receive audio signals from two separate audio sources and apply a MIDI based timing signature to the audio signals. The audio signals are synchronized according to the MTC signature signal and combined, when synchronized, at a summing module for output as a combined audio stream. Some embodiments include monitoring for and publishing a start/stop command signal to users in the collaboration to help synchronize audio signals from the different users.

Abstract: A storage controller coupled to a storage array comprising one or more storage devices performs a uniform sampling of data storage items in an append-only file system to identify a sample set and identifies at least two data storage item groups based on an age of the data storage items. The storage controller performs a garbage collection process on the append-only file system to identify stale data storage items in a first logical storage segment and active data storage items in the first logical storage segment and writes a first active data storage item associated with a first group of the two data storage item groups from the first logical storage segment to a second logical storage segment, the second logic storage segment comprising other active data storage items associated with the first group.

Abstract: Disclosed are an expression curve generating method based on voice input, and an device using the same. The method comprises the following steps: (1) inputting voice, and generating an audio file; (2) selecting expression graph buttons corresponding to a type of expressions from multiple expression graph buttons; (3) dividing the audio file into multiple equal time sections according to a time length, and respectively calculating a corresponding volume of each time section; (4) quantifying the corresponding volumes of the different time sections in the audio file into different volume levels; (5) obtaining expression icons corresponding to the volume level of each time section from a same group of expression icons, generating an expression curve using time as a horizontal axis and the volume level as a vertical axis; and (6) displaying the expression curve formed in step (5).

Abstract: A system is presented for generation of output sounds having psychoacoustic qualities comparable to input sound or sounds. Short term and intermediate term features are computed for each input sound, sound components are clustered, filtered, and scored; and a prediction learning system is trained on the probabilities of classes of regions over time. A decoder can make use of this information to generate outputs that sound similar to, but not the same as, the input sound or sounds. The method and apparatus can be operated with no special training.

Abstract: A sound effect parameter adjustment method, and a mobile terminal. The method comprises: obtaining a sound effect parameter of a mobile terminal and determining the sound effect parameter as an initial sound effect parameter; detecting, when playing an audio file by audio playback software, whether a playback sound effect parameter of the audio file is consistent with the initial sound effect parameter; if it is detected that the playback sound effect parameter of the audio file is not consistent with the initial sound effect parameter, determining whether the playback sound effect parameter of the audio file falls within a preset sound effect parameter interval; and if not, adjusting the sound effect parameter to be within the sound effect parameter interval.

Abstract: Disclosed in the present invention is a character curve generating method and device thereof. The method comprises the following steps: (1) inputting text information, and the text information comprises m characters; (2) performing voice input to generate an audio file; (3) dividing the audio file into m equal time periods and calculating a corresponding sound volume of each time period; (4) calculating dimension information corresponding to each character in the text information according to the sound volumes of different time periods in the audio file; and (5) changing display sizes of the m characters in the text information according to the dimension information acquired in step (4) to form a character curve. The text curve formed by multiple characters in different font sizes generates a more interesting image and provides more information compared to text information in a single font size in the existing instant communication system.

Abstract: An electronic musical instrument includes a plurality of keys respectively specifying different pitches when operated; a memory; and a sound processor. In response to a current operation of a current key, which is one of the plurality of keys, the sound processor retrieves the information stored in the memory for a previous operation, if any, of a previous key, which is a same as the current key or is another one of the plurality of keys, and performs a prescribed processing on a beginning part of the waveform data generated for the current operation of the current key in accordance with the retrieved information stored in the memory for the previous operation of the previous key so as to generate processed waveform data in response to the current operation of the current key. The resulting processed waveform data can be configured to better mimic artists' performance of an original instrument.

Abstract: A pitch detection method. Such a pitch detection method may use a Teager Energy Operator (TEO) with a Pseudo Weigner Ville Transformation (PWVT) to recover speech from noise and to recover low-frequency information of the speech signal in its detection of a pitch value. Also, the pitch detection method may use the combinatory PWVT and the respective state machine for decision making.

Abstract: Provided are apparatuses and methods in a digital broadcast transmission system for providing user interfaces to a mobile device. The user interfaces (e.g., “widgets”) may provide information to a mobile device user by displaying the interfaces in a predetermined location on a mobile device display. For example, the user interfaces may be displayed on a dashboard on a mobile device and may contain updated information. As information pertaining to content of the user interfaces changes, the status of the information in the user interfaces may change accordingly. In one example, an event user interface may provide information pertaining to the event and may be updated based on changes to the event. For example, the number of participants may change. This information may be provided in the user interface. In addition, user interfaces may be selected, modified, arranged, organized or otherwise manipulated at a server and downloaded to the mobile device.

Abstract: A content data generating device includes a first storage configured to store content data including at least either video information or audio information, a second storage configured to store variation data representing change of a parameter on the content data, a designator configured to designate a portion of the variation data, and a content data generator configured to process the content data according to a value of the parameter of the portion of the variation data designated by the designator to generate processed content data.

Abstract: A microchip has imbedded therein, circuitry for enhancing the sound outputted therefrom and which is associated with an electronic device, such as a radio, television, headphones, earbuds and the like.

Abstract: A method includes performing a harmonic decomposition on a target engine sound, thereby to extract each of N harmonics of the target engine sound over an RPM range; and out of the extracted harmonics, extracting phase and shape information for each of the N harmonics over the RPM range for reproducing the target engine sound. The method also includes configuring an engine harmonic enhancement (EHE) system to utilize the extracted phase and shape information to generate an engine harmonic enhancement signal, he(t), to be added on top of a baseline engine sound.

Abstract: An electric acoustic apparatus includes a supply unit that supplies sound-generation information including timing information representing a start of a sound-generation, a storage storing sound control pattern data representing a pattern of over-time change of a parameter defining a behavior of a sound, a reproducer that reproduces the sound control pattern data, and a sound signal generator that generates a sound signal representing a sound according to the sound-generation information and changes the behavior of the sound based on the reproduced sound control pattern data. The reproducer controls a reproduction behavior of the sound control pattern data according to the sound-generation information.

Abstract: A method and an apparatus for recording and evaluating a sound source are provided. The sound source evaluation method calculates a creativity index, which indicates a difference of a sound source property of test performance information with respect to a sound source property of reference performance information at a specific performing time, by comparing the test performance information, which is acquired by quantifying, according to passage of the performing time, a sound source property in a provided test sound source, with the reference performance information, which is acquired by extracting a sound source property and a sound source property having the same attribute according to passage of the performing time from a plurality of verified sound sources and then through a statistical analysis of the extracted properties.

Abstract: A session terminal apparatus constitutes a network session system together with another session terminal apparatus. In the session terminal apparatus, a back performance instruction part instructs reproduction of back performance information. A back performance control part transmits the back performance information, when the reproduction thereof is instructed by the back performance instruction part, to another session terminal apparatus, and controls a performance reproduction part to start reproducing the back performance information when a monitor delay time corresponding to a communication delay time between the session terminal apparatus and another session terminal apparatus has elapsed after the reproduction of the back performance information was instructed.

Abstract: A system for reducing noise in a sensor measurement system includes a noise detector for detecting noise in capacitive to digital conversion measurements; a noise correction module operably coupled to the noise detector and configured to dynamically modify one or more delays associated with a sampling waveform; and a noise filter for filtering the sampling waveform.

Abstract: In one embodiment, a programmable pedalboard for a musical instrument is provided. The pedalboard includes a docking station for receiving a removable portable computer that provides a plurality of instrument effects. The docking station is configured to receive an audio signal from a musical instrument and to modify the audio signal from the musical instrument based on at least one instrument effect from the plurality of instrument effects.

Abstract: Features are disclosed for performing efficient dereverberation of speech signals captured with single- and multi-channel sensors in networked audio systems. Such features could be used in applications requiring automatic recognition of speech captured with sensors. Dereverberation is performed in the sub-band domain, and hence provides improved dereverberation performance in terms of signal quality, algorithmic delay, computational efficiency, and speed of convergence.

Abstract: A power supply apparatus is provided. The power supply apparatus includes a power conversion circuit, an AC safety capacitor and a control unit. An AC input voltage is received by an input side of the power converting circuit, and is converted to a DC output voltage. The AC safety capacitor is connected across the input side. The control unit controls the operation of the power converting circuit, where the control unit samples and holds a voltage between two terminals of the AC safety capacitor to obtain a holding voltage, and compares the holding voltage and the AC input voltage during each period of a clock signal, so as to decide to discharge electric energy stored by the AC safety capacitor according to a comparison result.

Abstract: Systems and methods are provided for program identification. For example, a first audio fingerprint corresponding to a first audio signal is acquired; whether one or more second audio fingerprints in a predetermined fingerprint database match with the first audio fingerprint is detected, a second audio fingerprint corresponding to a second audio signal; and in response to one of the second audio fingerprints matching with the first audio fingerprint, a program associated with the matching second audio signal is provided as a result for program identification associated with the first audio signal.

Abstract: A method of producing dynamic controllable data composites from two or more data segments includes: building or training one or more function mappers to map between one or more extracted feature envelopes sets from the original data and one or more general parametric representations of the data; combining the extracted feature envelopes or the function mappers using two or more audio segments; and feeding the extracted feature envelopes or combined feature envelopes to the function mappers to obtain synthesis parameters to drive a synthesis process.

Abstract: Systems and methods for adjusting pitch of an audio signal include detecting input notes in the audio signal, mapping the input notes to corresponding output notes, each output note having an associated upper note boundary and lower note boundary, and modifying at least one of the upper note boundary and the lower note boundary of at least one output note in response to previously received input notes. Pitch of the input notes may be shifted to match an associated pitch of corresponding output notes. Delay of the pitch shifting process may be dynamically adjusted based on detected stability of the input notes.

Abstract: This disclosure details the implementation of apparatuses, methods and systems for audio processing and transmission. Some implementations of the system are configured to provide a method for encoding an arbitrary number of audio source signals using only a small amount of (transmitted or stored) information, while facilitating high-quality audio playback at the decoder side. Some implementations may be configured to implement, a parametric model for retaining the essential information of each source signal (side information). After the side information is extracted, the remaining information for all source signals may be summed to create a reference signal from which noise information for the original source signals may be reconstructed. The reference signal and the side information form the new collection of information to be transmitted or stored for subsequent decoding.

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: An information processing apparatus is provided which includes a music analysis unit for analyzing an audio signal serving as a capture source for a sound material and for detecting beat positions of the audio signal and a presence probability of each instrument sound in the audio signal, and a capture range determination unit for determining a capture range for the sound material by using the beat positions and the presence probability of each instrument sound detected by the music analysis unit.

Abstract: Herein provided is a system for singing synthesis capable of reflecting not only pitch and dynamics changes but also timbre changes of a user's singing. A spectral transform surface generating section 119 temporally concatenates all the spectral transform curves estimated by a second spectral transform curve estimating section 117 to define a spectral transform surface. A synthesized audio signal generating section 121 generates a transform spectral envelope at each instant of time by scaling a reference spectral envelope based on the spectral transform surface. Then, the synthesized audio signal generating section 121 generates an audio signal of a synthesized singing voice reflecting timbre changes of an input singing voice, based on the transform spectral envelope and a fundamental frequency contained in a reference singing voice source data.

Abstract: Systems and methods for selecting audio samples in response to musical stimuli are provided. In some embodiments, an audio sample can be selected based on the excitation state of an instrument. A musical stimulus can be received, and a current excitation level associated with previously received musical stimuli calculated. An audio sample can be selected for playback using the current excitation level. In some embodiments, audio samples having different velocity levels can be selected in response to repeated musical stimuli. A first instance of a musical stimulus having a first velocity level can be received, and a first audio sample corresponding to the first velocity level played back. A second instance of the musical stimulus having the first velocity level can be received, and a second audio sample corresponding to a second velocity level can be selected for playback. The first and second audio samples can have different audio characteristics.

Abstract: A sampling device includes: an obtainer that obtains a streaming audio waveform; a detector that detects a tap by a user; a designator that designates a sampling start point on the obtained audio waveform when at least a single tap has been detected by the detector, the designating being performed on the basis of one tap among a plurality of the taps when the plurality of taps are detected; a calculator that calculates a sampling duration on the basis of time intervals between the respective taps when the plurality of taps are detected, the calculating being performed from the sampling start point to a subsequent tap that is performed after the one tap; and a waveform sampler that samples the obtained audio waveform, the sampling starting from the designated sampling start point and ending in accordance with the calculated sampling duration.

Abstract: Embodiments of the present invention provide an electronic looping, recording, and playback apparatus. The apparatus may comprise a clamp, enabling the apparatus to be positioned in or on an acoustic instrument, such as an acoustic guitar. The clamp may be used to attach the apparatus to a portion of the acoustic instrument. The apparatus may further comprise, but not be limited to, for example, a speaker, a rechargeable battery, a microphone, a volume knob, an LED indicator light, a switch, and a processing module. The switch may be used to activate the electronic recording, looping, and playback system, much like a pedal in a stompbox configuration.

Abstract: In an ensemble performance, performance lesson or the like, a musical score (display content) is displayed and input audio is recorded. The input audio includes notes, comments or the like uttered by a human player or an instructor. During recording of the input audio is received a user input designating a desired time position on the musical score displayed on the display device. A recording time of the input audio based on a time point at which the user input has been received and the time position on the musical score designated by the user input are stored into a storage device in association with each other. An icon is displayed in association with the time position on the musical score designated by the user input. Once the icon is selected, voice based on the recording time recorded in association with the icon is reproduced to sound the comments, etc.

Abstract: Game systems according to various embodiments are adapted to allow users to assemble an instrument performance for a particular song by: (1) displaying a music loop pallet that includes a plurality of different music loops that each contain music played by a particular musical instrument (e.g., guitar, bass, drums, piano, etc.); and (2) allowing a user to assemble the instrument performance by indicating (a) which of the plurality of music loops will be included in the instrument performance, and (b) in which order the music loops will be played during the instrument performance. In certain embodiments, the system is configured to allow a user to create songs that comprise a plurality of instruments and genres of music into one complete song.

Abstract: A sound analysis and associated sound synthesis method is provided. A first input sound signal is received and analyzed, to determine its corresponding impulse response representative of a timbre of the input sound signal. A second input sound signal is received and processed into a form which the corresponding impulse response is susceptible to being applied, wherein the processing includes generating a “pink noise” equivalent frequency spectrum of the second input sound signal. The impulse response is applied to the processed second input sound signal to generate an output signal, wherein the output sound signal includes at least timbral nuances of the first input sound signal.

Abstract: This method of converting a displacement of a magnetic object into a signal directly perceptible by a human being, comprises: —the acquisition (32) of a temporal succession of measurements of a magnetic field modified by the displacements of the object during a sliding time window of predetermined duration, —the construction (34) on the basis of this acquired temporal succession of measurements of several signals each representing a characteristic of the measured magnetic field dependent on a corresponding characteristic of the displacement of the object, and —the adjustment (50) of several parameters of the directly perceptible signal as a function of the signals so as to render these characteristics directly perceptible.

Abstract: Systems and method for performing adaptive audio signal processing using music as a measurement stimulus signal. A musical stimuli generator may be used to generate musical stimulus signals composed to provide a stimulus with a spectrum that is substantially dense, and ideally white or pink, over a selected frequency range, so that all frequencies of interest are stimulated. The musical stimuli generator may generate melodically pleasing musical stimulus signals using music clips that include any of: a chromatic sequence, a chromatic sequence including chromatic tones over a plurality of octaves, a chromatic sequence including chromatic tones over a selected plurality of octaves, or an algorithmically composed chromatic sequence, to cover a selected frequency range. The musical stimulus signal may be generated as sound into the environment of use. An audio input picks up the sound from the environment, and a sound processor uses the received musical stimulus signal to determine a transfer function.

Abstract: A beat analysis module is described for determining beat information associated with an audio item. The beat analysis module uses an Expectation-Maximization (EM) approach to determine an average beat period, where correlation is performed over diverse representations of the audio item. The beat analysis module can determine the beat information in a relative short period of time. As such, the beat analysis module can perform its analysis together with another application task (such as a game application task) without disrupting the real time performance of that application task. In one application, a user may select his or her own audio items to be used in conjunction with the application task.

Abstract: Dampers provided in corresponding relation to a plurality of keys are each controlled in response to both an operation of a pedal and an operation of the corresponding key. One half region or half point is determined based on a plurality of half pedal regions or half pedal points, in a stroke of the pedal, specific to the individual dampers. Performance data instructing a pedal operation is generated and recorded on the basis of a stroke position detected in response to a pedal operation and the determined one half region or half point. For each of the keys, a key-damper half region or key-damper half point in a stroke of the key is identified in advance. Performance data instructing key operations is generated and recorded on the basis of the key-damper half regions or key-damper half points specific to the keys and stroke positions detected in response to key operations.

Abstract: One half region or point is determined based on a plurality of half pedal regions or points, in a stroke of a pedal, specific to the individual dampers corresponding to keys. When a pedal reproduction event instructs a half region or point, a target trajectory of the stroke of the pedal is generated such that the pedal is positioned at the half region or point specific to the key, and the pedal is driven on the basis of the target trajectory. For each key, a key-damper half region or point in a stroke of the key is identified in advance. In response to a key reproduction event instructing half control, a target trajectory of the stroke of the key is generated such that the key is positioned at the key-damper half region or point specific to the key, and the key is driven based on the target trajectory.

Abstract: A system and method is disclosed that facilitates the processing of a sound signal. In embodiments, an input sound signal can be processed according to a computational model using predetermined parameters. A sound signal originating from a musical instrument can be processed according to coefficients that are generated using a learning model.

Abstract: Systems and methods of synchronizing media are provided. A client device may be used to capture a sample of a media stream being rendered by a media rendering source. The client device sends the sample to a position identification module to determine a time offset indicating a position in the media stream corresponding to the sampling time of the sample, and optionally a timescale ratio indicating a speed at which the media stream is being rendered by the media rendering source based on a reference speed of the media stream. The client device calculates a real-time offset using a present time, a timestamp of the media sample, the time offset, and optionally the timescale ratio. The client device then renders a second media stream at a position corresponding to the real-time offset to be in synchrony to the media stream being rendered by the media rendering source.

Abstract: A method of performing audio synthesis is disclosed. An audio event is input to an audio algorithm along with associated parameters including source sample data. An interpolation function is provided and the source sample data are interpolated to generate one or more interpolated samples based on the source sample data. A filter function is provided and at least one of the interpolated samples is filtered to generate a filtered sample. A gain function is provided and the filtered sample is processed to generate a gained sample. At least one of the interpolation, filter, and gain functions include outputting an earlier-calculated value along with an estimated difference value in lieu of calculating a new value.

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 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.