Abstract: Embodiments of the present invention provide a power supply unit capable of achieving noise reduction while maintaining efficiency under the light load state. A FET driver controls switching elements in either one of a Pulse Width Modulation (PWM) mode, an intermittent mode having a lower operating frequency than that in the PWM mode, and a noise-free mode having a higher operating frequency than an audible frequency range. The FET driver operates first in the intermittent mode under the light load state. A microphone collects noise generated from the surroundings of a power supply unit. When the level of an audio signal collected by the microphone exceeds a predetermined level, the FET driver transitions from the intermittent mode to the noise-free mode. In accordance with such an embodiment, the FET driver operates in the noise-free mode only when noise is actually generated.

Abstract: An audio signal processing system is configured to separate an audio signal into a dry signal component and one or more reverberant signal components. The dry signal component and the reverberant signal components can be separately modified and then recombined to form a processed audio signal. Alternatively, the dry signal component may be combined with an artificial reverberation component to form the processed audio signal. Modification of the reverberation signal component and generation of the artificial reverberation component may be performed in order to modify the acoustic characteristics of an acoustic space in which the audio signal is driving loudspeakers. The audio signal may be a pre-recorded audio signal or a live audio signal generated inside or outside the acoustic space.

Abstract: A hand-held music player for use in conjunction with radios, including a casing, a receiver socket on the casing through which digital audio data is received, a digital-to-analog audio converter housed within the casing, a first transfer socket on the casing through which a song is transferred to a radio transmitter, a second transfer socket on the casing through which meta-data for the song is transferred to the radio transmitter, and a dial on the casing for selecting a song for playback. A method and a computer-readable storage medium are also described.

Abstract: A method and system is provided for eliminating acoustical feedback in a system. The method determines a parameter for at least one notch filter, adjusting the notch filter based on the parameter, processing the digital signals through the notch filter, testing at the effect of the notch filter in the system, and removing the notch filter if the notch filter is not effective. Also disclosed is a method and system of selecting candidate frequencies which might be feedback, as opposed to other wanted sound frequencies. The selection method sampling the digital signals, converting the time domain digital signal samples by a fast Fourier transform algorithm into the frequency domain, using a ballistics approach to find prominences in the frequency spectrum, and testing the sizes of the prominences.

Abstract: Exemplary embodiments of methods and apparatuses to correlate changes in one audio signal to another audio signal are described. A first audio signal is outputted. A second audio signal is received. The second audio signal may be stored in a memory buffer. The first audio signal is correlated to conform to the second audio signal. The first audio signal may be dynamically correlated to match with the second audio signal while the second audio signal is received. At least in some embodiments, a size of a musical time unit of the second audio signal is determined to correlate the first audio signal. At least in some embodiments, the adjusted first audio signal is stored in another memory buffer.

Abstract: A method and system is provided for eliminating acoustical feedback in a system. The method determines a parameter for at least one notch filter, adjusting the notch filter based on the parameter, processing the digital signals through the notch filter, testing at the effect of the notch filter in the system, and removing the notch filter if the notch filter is not effective. Also disclosed is a method and system of selecting candidate frequencies which might be feedback, as opposed to other wanted sound frequencies. The selection method sampling the digital signals, converting the time domain digital signal samples by a fast Fourier transform algorithm into the frequency domain, using a ballistics approach to find prominences in the frequency spectrum, and testing the sizes of the prominences.

Abstract: A system, device, and method for recording audio with the character and sonic benefits of a genuine analog recording is disclosed. More specifically, an electro-mechanical-software controlled closed loop analog signal processor (“CLASP”) system which is comprised of a CLASP unit containing firmware, a latency detection module, and CLASP hardware display and controls. The CLASP system further comprises CLASP software operably running on a digital audio workstation (“DAW”) which is also in operable communication with the CLASP unit. The CLASP unit is also in operable communication with an analog recordable medium. An analog audio signal is recorded on the analog recordable medium, which may consist of a coated tape, cup, cylinder, drum, or disk, and then immediately played back and routed to the DAW via an analog to digital converter, thus providing for digitally recorded analog audio. The CLASP system may also include converters and a mixing console.

Abstract: A transient detector is provided for generating an ambience signal suitable for being emitted via loudspeakers for which there is no special loudspeaker signal to detect a transient period. A synthesis signal generator produces a synthesis signal which fulfills the transient condition on the one hand and the continuity condition for the synthesis signal on the other hand. A signal substituter will then substitute a portion of the examination signal by the synthesis signal to obtain an ambience signal for the surround channels.

Abstract: A signal processing apparatus includes a source separation module for producing respective separation signals corresponding to a plurality of sound sources by applying an ICA (Independent Component Analysis) to observation signals produced based on mixture signals from the sound sources, which are taken by source separation microphones, to thereby execute a separation process of the mixture signals, and a signal projection-back module for receiving observation signals of projection-back target microphones and the separation signals produced by the source separation module, and for producing projection-back signals as respective separation signals corresponding to the sound sources, which are taken by the projection-back target microphones. The signal projection-back module produces the projection-back signals by receiving the observation signals of the projection-back target microphones which differ from the source separation microphones.

Abstract: This invention describes a method for adjusting the loudness and the spectral content of digital audio signals in a real-time using warped spectral filtering. A warped processing module modifies a spectral content of a digital audio signal with a set of gains for a plurality of non-linearly-scaled frequency bands determined by a warping factor ? of a warped delay line. Warped delay line signals, generated by the warped delay line, are processed by a warped filter block containing multiple warped finite impulse response filters, e.g., Mth band filters, using individual warped spectral filtering in said plurality of the non-linearly-scaled frequency bands, which is followed by a conventional processing by a dynamic range control/equalization block. The present invention describes another innovation, that is embedding the warped processing module in a two-channel quadrature mirror filter (QMF) bank for improving processing efficiency at high sample rates.

Abstract: Every extreme value in an audio waveform represented by a digital audio signal having a sequence of samples is detected. A number of samples between samples corresponding to the first and second latest extreme values is detected. A corrective value is generated in response to the detected sample number and a difference between the first and second latest extreme values. Ones are designated among samples in response to the detected sample number. The designated samples include at least (1) a sample adjacently following the sample corresponding to the second latest extreme value, (2) a sample adjacently preceding the sample corresponding to the first latest extreme value, and (3) one of the sample corresponding to the first latest extreme value and the sample corresponding to the second latest extreme value. The designated samples are corrected in response to at least one of current, previous, and feature corrective values.

Abstract: Simple, computational efficient, and robust audio features are applied in a uniform program indexing method for picking up video segments relating to highlight plays in a recorded program worthy of being reviewed. By focusing on certain frequencies in an audio sequence of the program, a computational complexity of the uniform program indexing method is significantly decreased. With the aid of MFCC coefficients and a DFBE coefficient generated from the MFCC coefficients, audio patterns may be utilized for differentiating exciting events in the program from other unnecessary information. Scores corresponding to various audio segments are regarded as standards for picking up video segments in the program worthy of being chosen in a recorded highlight collection. Some low-level-feature parameters, some video segments having highlight-related visual characteristics, and a re-ranking procedure are utilized for enhancing precision of the scores for providing video segments worthy of being reviewed.

Abstract: Character value of a sound signal is extracted for each unit portion, and degrees of similarity between the character values of the individual unit portions are calculated and arranged in a matrix configuration. The matrix has arranged in each column the degrees of similarity acquired by comparing, for each of the unit portions, the sound signal and a delayed sound signal obtained by delaying the sound signal by a time difference equal to an integral multiple of a time length of the unit portion, and it has a plurality of the columns in association with different time differences. Repetition probability is calculated for each of the columns corresponding to the different time differences in the matrix. A plurality of peaks in a distribution of the repetition probabilities are identified. The loop region in the sound signal is identified by collating a reference matrix with the degree of similarity matrix.

Abstract: A user-friendly system for real time performance and user modification of one or more previously recorded musical compositions facilitates user involvement in the creative process of a new composition that reflects the user's personal style and musical tastes. Such a system may be implemented in a small portable electronic device such as a handheld smartphone that includes a stored library of musical material including original and alternative versions of each of several different components of a common original musical composition, and a graphic user interface that allows the user to select at different times while that original composition is being performed, which versions of which components are to be incorporated to thereby create in real time a new performance that includes elements of the original performance, preferably enhanced at various times with user selected digital sound effects including stuttering and filtering.

Abstract: The MPEG2 Advanced Audio Coder (AAC) standard limits the number of filters used to either one filter for a “short” block or three filters for a “long” block. In cases where the need for additional filters is present but the limit of permissible filters has been reached, the remaining frequency spectra are simply not covered by TNS. Two solutions are proposed to deploy TNS filters in order to get the entire spectrum of the signal into TNS. The first method involves a filter bridging technique and complies with the current AAC standard. The second method involves a filter clustering technique. Although the second method is both more efficient and accurate in capturing the temporal structure of the time signal, it is not AAC standard compliant. Thus, a new syntax for packing filter information derived using the second method for transmission to a receiver is also outlined.

Abstract: A method for recording and replaying operations in a computer environment utilizes initial conditions of the computer environment at the start of a recording to configure a replay computer environment during replay. The initial conditions of the computer environment are saved prior to recording of user inputs to the computer environment. The saved initial conditions and the recorded user inputs can then be used to actively operate the replay computer environment from a state substantially identical to the initial state of the computer environment to replay the recorded operations in the replay computer environment. The method also includes a technique to synchronize the operations with accompanying audio during replay.

Abstract: Tools and techniques are provided to allow the user of a signal editing application to retain control over individual changes, while still relieving the user of the responsibility of manually identifying problems. Specifically, tools and techniques are provided which separate the automated finding of potential problems from the automated correction of those problems. Thus, editing is performed in two phases, referred to herein as the “analysis” phase and the “action” phase. During the analysis phase, the signal editing application automatically identifies target areas within the signal that may be of particular interest to the user. During the “action” phase, the user is presented with the results of the analysis phase, and is able to decide what action to take relative to each target area.

Abstract: A sound reproduction and amplification system includes a digital central controller, a wireless transmitter and a plurality of addressable wireless digital receivers and digital amplifiers for driving loudspeakers or earphones, wherein Differential Pulse Width Modulation (DPWM) signals from the central control of the audio transmitter are sent to the addressable receivers, but no DPWM signals are sent unless there are changes in the target PWM signals. The control signaling is based on position mapping in each repetitive sequence of bits (i.e., each frame or word) in a digital communication channel, where only a single bit per channel per word is allotted to each receiver/amplifier/loudspeaker. If there is any change in output of any transmitter PWM from the audio processor (decoder), all the channel bits are sent to all the addressable loudspeakers.

Abstract: An apparatus and method for preparing a playlist in a sound source data player are provided. For the preparation of the playlist, the position of a sampling of the sound source data is set by user's arbitrary selection or by searching out the thematic part of the sound source data. Then, if the sampling time taken for playing back the sampling has been set, the player plays back the sound source data file selected by moving a cursor from the position of the sampling. Finally, the user adds the individual information of the presently selected sound source data to the playlist. Thus, the user may quickly and readily sample a large number of songs stored in the player and thus more easily prepare the playlist.

Abstract: A method and system for transcoding for a mobile device. The method includes accepting a master file, processing the master file with steps laid out in a definition file, the definition file providing instructions for converting the master file to a derivative file appropriate for playback on a pre-specified mobile handset, and outputting the results of the processing step into the derivative file. The system includes a definition file, the definition file providing instructions for converting a master file to a derivative file appropriate for playback on a pre-specified mobile handset. A plurality of modules is employed that are controlled by the definition file and which perform a corresponding plurality of functions on the master file or derivatives thereof.