56 results about "Programming" patented technology

A digital wavetable audio synthesizer is described. The synthesizer can generate up to 32 high-quality audio digital signals or voices, including delay-based effects, at either a 44.1 KHz sample rate or at sample rates compatible with a prior art wavetable synthesizer. The synthesizer includes an address generator which has several modes of addressing wavetable data. The address generator's addressing rate controls the pitch of the synthesizer's output signal. The synthesizer performs a 10-bit interpolation, using the wavetable data addressed by the address generator, to interpolate additional data samples. When the address generator loops through a block of data, the signal path interpolates between the data at the end and start addresses of the block of data to prevent discontinuities in the generated signal. A synthesizer volume generator, which has several modes of controlling the volume, adds envelope, right offset, left offset, and effects volume to the data. The data can be placed in one of sixteen fixed stereo pan positions, or left and right offsets can be programmed to place the data anywhere in the stereo field. The left and right offset values can also be programmed to control the overall volume. Zipper noise is prevented by controlling the volume increment. A synthesizer LFO generator can add LFO variation to: (i) the wavetable data addressing rate, for creating a vibrato effect; and (ii) a voice's volume, for creating a tremolo effect. Generated data to be output from the synthesizer is stored in left and right accumulators. However, when creating delay-based effects, data is stored in one of several effects accumulators. This data is then written to a wavetable. The difference between the wavetable write and read addresses for this data provides a delay for echo and reverb effects. LFO variations added to the read address create chorus and flange effects. The volume of the delay-based effects data can be attenuated to provide volume decay for an echo effect. After the delay-based effects processing, the data can be provided with left and right offset volume components which determine how much of the effect is heard and its stereo position. The data is then stored in the left and right accumulators.

Disclosed herein are systems and methods enabling hearing aid fitting by a non-expert consumer. The method in one embodiment involves delivering a sequence of test audio signals corresponding to natural sound segments to a non-acoustic input of a programmable hearing device in-situ, while allowing the consumer to adjust fitting parameters based perceptual assessment of hearing device output. The sound segments define a fitting soundscape within the normal human auditory range, with each sound segment corresponding to one or more fitting parameters of the programmable hearing device. The consumer is instructed to listen to the output of the in-situ hearing device and adjust controls related to corresponding fitting parameters. In one embodiment, the fitting system comprises a personal computer and a handheld device providing calibrated test audio signals and programming interface. The systems and methods allow home dispensing of hearing devices without requiring specialized instruments.

Systems and methods for 3D audio programming and processing are provided wherein digital signal processing (DSP) settings are calculated for 3D audio effects for a digital audio signal independently of DSP rendering of the digital audio signal. Coordinates of locations within a 3D environment representing at least one sound source and at least one audio listener are created and passed (along with other distance modeling parameters) to a DSP settings generator having 3D audio library routines to calculate the DSP settings for 3D audio effects based on the distances between the at least one sound source and at least one audio listener and the distance modeling parameters. Support for positional multi-channel sounds and non point-source emitters is also present.

A method including the steps of assessing at least two metadata parameters associated with an audio bitstream (e.g., an encoded Dolby Digital (AC-3), Dolby Digital Plus, or Dolby E bitstream), determining individual metadata parameter quality values, each of the individual metadata parameter quality values indicative of quality (e.g., correctness) of a different one of the at least two metadata parameters, and generating data indicative of a metadata score, where the metadata score is a value determined by a combination (e.g., a linear combination or other weighted combination) of the individual metadata parameter quality values. The metadata score is indicative of overall quality (e.g., correctness) of the at least two metadata parameters. Another aspect is a system (e.g., a test device or measurement device, or another test or measurement product, or a processor) configured (e.g., programmed) to perform any embodiment of the method.

Microprocessor technology is used to detect routine sounds in a substantially enclosed environment to determine normal or abnormal activity or noise within the environment (i.e., habitual behavior of an individual) A device is situated in a substantially enclosed environment with audio input device similar to a microphone and an optional visual display for interaction with the local users. The device has the ability to be controlled locally via key-pad or USB port connected to a local laptop computer, or remotely via a phone line or Ethernet connection to internet. The device further performs audio pattern recognition using waveform matching scheme to detect the occurrence of pre-programmed sounds representing routine activities. The microprocessor counts the number occurrence of each recognizable sound for a particular interval and over a length of a day or other duration and reports to a remote server. The remote server performs tracking and trending analysis to a web-based caregiver client. Significant changes are detected and reported to caregiver or family web-based client.

For courses that deal with media content, such as sound, music, photographic images, hand sketches, video, conventional techniques for automatically evaluating and grading assignments are generally ill-suited to direct evaluation of coursework submitted in media-rich form. Likewise, for courses whose subject includes programming, signal processing or other functionally-expressed designs that operate on, or are used to produce media content, conventional techniques are also ill-suited. Instead, it has been discovered that media-rich, indeed even expressive, content can be accommodated as, or as derivatives of, submissions using feature extraction and machine learning techniques. In this way, e.g., in on-line course offerings, even large numbers of students and student submissions may be accommodated in a scalable and uniform grading or scoring scheme. Likewise, large collections of coursework submissions (whether or not graded or scored) or media content more generally, may be efficiently browsed and grouped using techniques described herein.

In some embodiments, a method for adaptive control of gain applied to an audio signal, including steps of analyzing segments of the signal to identify audio objects (e.g., voices of participants in a voice conference); storing information regarding each distinct identified object; using at least some of the information to determine at least one of a target gain, or a gain change rate for reaching a target gain, for each identified object; and applying gain to segments of the signal indicative of an identified object such that the gain changes (typically, at the gain change rate for the object) from an initial gain to the target gain for the object. The information stored may include a scene description. Aspects of the invention include a system configured (e.g., programmed) to perform any embodiment of the inventive method.

A wearable apparatus for processing an audio signal may include at least one microphone configured to capture the audio signal from an environment of a user of the wearable apparatus and at least one processor. The at least one processor may be programmed to analyze the audio signal to identify an audio trigger. After identifying the audio trigger, the at least one processor may be programmed to store a portion of the audio signal containing a target audio segment related to the audio trigger and determine an action by analyzing the portion of the audio signal.

The invention provides a software framework that allows real-time computer-generated music to be used in interactive applications, particularly video games, by “modularizing” music-producing and music-modifying computer procedures into musically logical and programmatically convenient structures, as well as providing a communication mechanism between the application and the music-generating system which will allow the music to reflect the appropriate mood given the current application state.

A system for delivering audio, visual, or audiovisual entertainment, educational material and advertising to users of public and private restrooms, including visual images, and or sound programming, throughout the restroom or in discrete areas thereof, placed so that the images and / or sound can be conveniently seen and / or heard by users of these facilities. Optionally the system may be programmed to turn on when the restroom facilities are being used, and to be turned off either when the facility is vacated, or after a set time has elapsed.

The invention discloses an audio noise reduction accelerator system and method based on RISC v custom instruction set expansion, and belongs to the technical field of integrated circuits. The system mainly comprises an E203_CORE, an NICE_CORE, a NICE_Interface, an E203_SOC, an audio coding and decoding WM8731 module, and an audio noise reduction FxLMS algorithm, wherein the E203_CORE is connected with the NICE_CORE through the NICE_Interface, the E203_CORE, the NICE_CORE and a related peripheral port form the E203_SOC together, the E203_SOC is connected with the audio coding and decoding WM8731 module, and the audio noise reduction FxLMS algorithm is downloaded into a core of the RISC v processor through software programming to run. Compared with a processor of an ARM instruction set architecture, the processor adopting an RISC v custom instruction set can accelerate a specific operation part in the audio noise reduction FxLMS algorithm; and the problems of area, power consumption, granularity and the like can be better optimized, and the flexibility and feasibility of the algorithm are improved.

A system for documenting the music learning process of a music student to encourage more students to study long enough to become proficient in their musical form through involvement with an immersive interactive tool. A server computer is programmed to facilitate communication between the music student and other individuals. A music student computer is connected to the server computer. The music student computer is for transmitting and receiving music information related to the learning process of the music student. Likewise, a third party computer is connected to the server computer. The third party computer allows a pre-approved third party to transmit and receive music information related to the learning process of the music student An administrator computer is also connected to the server computer. The administrator computer permits an administrator to coordinate communication between the music student and the third party. In a preferred embodiment the third party computer includes friend and family member computers connected to the server computer. These friend and family member computers are for transmitting and receiving music information related to the learning process of the music student. Preferably, the music information related to the learning process of the music student includes photos, audio files and video files.

The invention discloses a control system, which can realize "freehand sketching type fast lamplight programming control and lamp layout control", "control of lamplight programs by a player and synchronous operation of audio and video files" and "synchronous trigger of the lamplight programs by audio and video signals". The control system comprises a central processor, a touch screen, a lamp function selection module, a storage unit, a control output interface, an audio signal receiving unit, an audio signal processing unit, a peripheral communication interface and a lamp, the touch screen, the lamp function selection module, the storage unit, the control output interface, the audio signal processing unit and the peripheral communication interface are in instruction communication connection with the central processor separately, the audio signal receiving unit is in communication connection with the audio signal processing unit, and the control output interface is electrically connected with the lamp. The control system disclosed by the invention has the advantages of simple operation and high playing property and high degree of freedom.

A portable streaming audio player and methods of using and programming the portable streaming audio player are described. The portable streaming audio player can include an interface panel that includes one or more assignable selection buttons to which streaming audio content can be assigned.

A vending machine and vending method for audio, video, music, movie or other date file downloads is optimized for commercial viability and product placement. The vending machine offers standard connectors allowing users to access a library of download files and download those files to data players such as MP3 music download players, video players and so on. It further offers inexpensive data players to users in addition to offering downloads. Importantly, the device has hardware or software (video) softbuttons allowing remote reprogramming: in the product placement method of the invention, a commercial entity may pay to have the softbutton programmed to their desired data file source. Commercial entities buying placement on the vending machine may be artists, performers, studios, recording labels, on-line and other music libraries. The machine may accept coins, thus increasing saleability of the data products sold thereon.

The invention relates to the field of music instrument making, in particular to an automated string production device and method. The device comprises a rack, a working platform arranged on the rack, core wire high-speed rotating devices, a mobile load platform, a wire discharge device arranged on the mobile load platform, a tensioning adjusting mechanism, a wire feeding mechanism, a control system, a UI interface and a control area. By means of the automated string winding and polishing equipment, semi-automation winding processing of strings is achieved, high-precision machining of multiple strings with different types and different specifications is achieved in a digital control programming mode, different process modes can be output according to different strings, and multiple purposes are achieved through one machine; compared with an original artificial winding processing mode, the processing efficiency is improved, reliable process quality is achieved, by means of the automatic polishing function, manpower can be saved, the problem that the quality is out of control due to artificial operation fatigue and other problems can be avoided, and the equipment is simple in integral structure and convenient to adjust and maintain.

An Audio streaming player, which can play the audio streaming by connecting to the Internet with a wired or wireless interface, the Audio streaming player can connect the audio streaming provider servers and play the audio streaming provided by these servers; the audio streaming player is composed with a programmable components for updating the codec and software with each particular audio streaming file format, and the behaviors control of this device is also pre-programmed; Moreover, the processing unit is for calculate the decode or code scheme and output a digitalized signals, besides the memory unit of this device provides the buffering space and storage place for the usage when decoding a audio stream, storing a address list server or other memorize requests; users can play the audio streaming by the cooperation of programmable components, memory units, digital signal processors, and Input / Output interfaces that belongs to this device.

The invention belongs to the technical field of infrared remote controllers, and relates to a general programmable remote controller and a programming method thereof. The general programmable remote controller is characterized in that the general programmable remote controller includes a core processing unit, an internal or external storage unit, and an audio signal processing unit, wherein the audio signal processing unit is responsible for shaping and amplification of an audio signal; the core processing unit is responsible for receiving an audio signal, decoding, carrying out read-write access on the storage unit, carrying out inputting treatment on a keyboard and transmitting infrared coded data; and the storage unit is responsible for storing a code number, a remote controller model, client information, remote control coding and button data. The programming method of the general programmable remote controller is characterized in that mobile phone software, flat computer software or computer software carrying the code number, the type, the brand and the model of an electric appliance, remote control coding and button data programs the remote controller data into the remote controller through an audio port by adopting an audio line through the method.

The embodiment of the invention provides an audio and video data stream processing method and system, and the method comprises the steps that an articulated naturality web terminal transmits request information of an audio and video data stream to an Ethernet server; the articulated naturality web terminal receives an audio and video data stream returned by the Ethernet server according to the serial number of the audio and video data stream in the request information; the articulated naturality web terminal carries out decoding operation on video data streams in the audio and video data streams through a built-in first application programming interface to obtain original video data, and carries out decoding operation on audio data streams in the audio and video data streams through a built-in second application programming interface to obtain original audio data; and the articulated naturality web terminal synchronously plays the original video data and the original audio data according to the timestamp in the original video data and the timestamp in the original audio data. According to the embodiment of the invention, the function that the articulated naturality web terminal canplay the audio and video data streams on the Ethernet terminal in the articulated naturality web is realized.

An exemplary programming system 1) detects a first input command representative of a request to create a sound processing program, 2) provides, in response to the first input command, a user interface that shows a plurality of default values corresponding to a plurality of parameters associated with the sound processing program, 3) detects a changing, by way of the user interface, of the plurality of default values to a plurality of modified values, 4) detects, subsequent to the changing, a second input command representative of a request to create a sound processing program template configured to serve as a basis for one or more additional sound processing programs, and 5) presents, in response to the second input command, an option that allows the user to select one or more of the modified values to be included in the sound processing program template.

A guiding device for lection reading and the relevant method is provided, which comprises a display, a microprocessor, a clock oscillator, storage, a phonetic integrated circuit, a microphone, a loudhailer, an audio output port, a control button and a volume control button. The display itself is equipped with a display driver while the phonetic integrated circuit is composed of an audio input interface, an audio output interface, a multi-section phonetic storage and an audio signal amplifier. The method of the lection reading under guidance is that: firstly, after digital compression, respectively pre-storing four groups of the lection and the audio prompts of beginning, intermediate and advanced levels, which are combined with date conditions of each week by software programming, into the multi-section phonetic storage inside the phonetic integrated circuit. Twelve different levels of the phonetic prompts are combined with the lection content to guide the user to read the lections. With the lection-reading guiding device by direct audio prompt which is simply controlled and conveniently operated, the said device and method simplify the lection reading and learning for different users.

The invention relates to an intelligent lamplight acousto-optic linkage control system and a control method thereof and relate to the technical field of acousto-optic linkage. The control system comprise a lamp used for indoor illumination and rendering, an audio acquisition module used for carrying out sampling and logical operation processing on audio signals and classifying the frequencies of the audio signals, a material module used for providing data packets for compiling various scenes of the lamp and the audio acquisition module, a storage module used for storing data information of each module, and a control unit module which is connected with the lamp, the audio acquisition module and the material module, and performs calculation, logic and control output to realize acousto-opticlinkage. The intelligent controller of the control system is high in integration level; lamplight can be changed along with audios, rhythms, beats, drumheads and the like of music, an acousto-optic linkage mode is achieved, a voice control acceleration function is achieved, an overall effect is good, and programming is simple.

The invention discloses an egg quality nondestructive detection device and method based on a magnetostriction technology, and relates to the technical field of egg detection. The device comprises a hollow shell, a magnetostrictive actuator, an audio power amplification circuit, a rubber pad, a microphone, a computer and an egg detection software system. According to the device, a computer software programming control sound card generates swept frequency excitation on the magnetostrictive actuator so that eggs placed on the magnetostrictive actuator can generate vibration audio signals, the microphone receives the vibration audio signals of the eggs and transmits the vibration audio signals into the computer in real time, and computer software performs egg quality analysis according to the vibration audio signals. According to the method, egg quality information in the audio signals is enhanced through a swept frequency vibration method, and in combination with a signal processing and mode recognition method, nondestructive detection of egg quality is achieved. The egg quality nondestructive detection device and method based on the magnetostriction technology can be applied to nondestructive detection of egg quality and particularly can achieve high-precision detection of eggs with fine cracks.

The invention discloses a method and a system for diagnosing a transformer fault by using sound feature coding. The method comprises the following steps: acquiring a transformer sound sample; extracting a plurality of sound features of the transformer sound sample; for each sound feature, obtaining a corresponding feature code according to a preset transformer sound feature coding rule table, and combining the feature codes of all the sound features to form a feature code combination; and querying a preset transformer fault table according to the feature code combination to obtain a transformer fault diagnosis result. The method and the system for diagnosing the transformer fault by using sound feature coding have the advantages of simple diagnosis method, easy programming realization, multiple diagnosis fault types and high diagnosis accuracy.

System and methods for processing audio signals are disclosed. In one implementation, a system may include at least one microphone configured to capture sounds from an environment of a user; and at least one processor. The processor may be programmed to receive at least one audio signal representative of at least part of the sounds captured by the microphone; identify at least one word in the at least one audio signal; and in response to identifying the at least one word, cause feedback to be provided the user.