Abstract: A composite variable duration soundtrack for a user to play while reading a text source, the soundtrack duration being defined by a soundtrack timeline. The soundtrack comprises multiple sound layers configured to play concurrently through the soundtrack timeline, each sound layer having an arrangement of one or more audio features that are configured to play at preset start times in the soundtrack timeline. At least one sound layer is adapted for modifying the preset start and stop times of its audio features to match the reading speed of a user based on a reading speed input.

Abstract: Uses of an enhanced sidetone signal in an active noise cancellation operation are disclosed. In one example, a method of audio signal processing includes producing an anti-noise signal based on information from a first audio signal. A target component of a second audio signal is separated from a noise component of the second audio signal to produce at least one among a separated target component and a separated noise component. Based on at least one among the separated target component and the separated noise component, an audio output signal is produced.

Abstract: An audio media visualization method and system. The method includes receiving by a computing processor, mood description data describing different human emotions/moods. The computer processor an audio file comprising audio data and generates a mood descriptor file comprising portions of the audio data associated with specified descriptions of the mood description data. The computer processor receives a mood tag library file comprising mood tags mapped to animated and/or still objects representing various emotions/moods and associates each animated and/or still object with an associated description. The computer processor synchronizes the animated and/or still objects with the portions of said audio data and presents the animated and/or still objects synchronized with the portions of said audio data.

Abstract: A method of creating a file according to a freeview audio/video (AV) service, the method including: creating a file header that includes content related information, AV object related information, preset related information, or preset information associated with at least one preset; creating audio object information associated with at least one audio object; creating video object information associated with at least one video object; and processing the file header, the audio object information, and the video object information into a preset format of a single freeview AV file to store the processed preset format of the freeview AV file is provided.

Abstract: A copyright protection method of audio data applied to an electronic device. Left and right channel audio signal values are retrieved from audio signals of an audio source. Enveloping difference values between each left channel audio signal and each right channel audio signal are calculated to determine a time slot. The left channel audio signals and the right channel audio signals respectively modulated, thereby writing digital copyright information in corresponding positions of the time slot according to the modulation.

Abstract: A single enclosure loudspeaker system projects multi-channel surround sound into a listener's room, and so replaces multiple conventional surround channel loudspeakers. The loudspeaker system includes a pair of opposing multi transducer arrays oriented laterally toward walls or reflecting surfaces (relative to the viewing axis) within the media space. The multi-element arrays are housed in a single self-powered loudspeaker enclosure along with a single (or multiple) low-frequency electro-acoustical drive element(s). In one embodiment of the invention, surround channel program material is pre-processed by an integrated wireless transmission device that performs certain digital signal processing and channel mixing steps in advance of wireless surround signal broadcast to a receiver.

Abstract: A noise cancelling device includes an extracting unit configured to extract a first noise from a signal, the signal being based on an input audio signal, a storing unit configured to store noise characteristic information on a second noise, the second noise remaining after subtracting the extracted first noise from the signal based on the audio signal. And the device further includes a cancelling unit configured to perform cancelling processing for cancelling a noise on the input audio signal based on the first noise and the noise characteristic information on the second noise.

Abstract: A miniature electronic shotgun microphone, which is used to receive a sound source from a specified direction, comprises a pick-up member, an A/D (Analog/Digital) conversion unit, and a digital signal processor. The pick-up member includes a first pick-up unit, a second pick-up unit separated from the first pick-up unit by a first distance, and a third pick-up unit separated from the second pick-up unit by a second distance; the first distance is greater than the second distance. The first pick-up unit, the second pick-up unit and the third pick-up unit respectively receive the sound source and output an analog signal. The A/D conversion unit and the digital signal processor process the analog signals, and convert them into a directional digital acoustic signal. Thus, the directional digital acoustic signal has a maximum pick-up frequency. Thereby is decreased grating lobes and spatial aliasing.

Abstract: A composite variable duration soundtrack for a user to play while reading a text source, the soundtrack duration being defined by a soundtrack timeline. The soundtrack comprises multiple sound layers configured to play concurrently through the soundtrack timeline, each sound layer having an arrangement of one or more audio features that are configured to play at preset start times in the soundtrack timeline. At at least one sound layer is adapted for modifying the preset start and stop times of its audio features to match the reading speed of a user based on a reading speed input.

Abstract: A POP noise suppression circuit includes a power source terminal, a clock signal input terminal, a charge unit, a discharge unit, a common-mode voltage judging and switching control unit, a charge and discharge capacitor, and a ground terminal. The charge unit includes a first clock generation circuit for generating a first pair of non-overlapped clock signal, and a first equivalent resistor. The discharge unit includes a second clock generation circuit for generating a second pair of non overlapped clock signals, and a second equivalent resistor. The charge unit generates a charge voltage changing slowly to the charge and discharge capacitor. The discharge unit generates a discharge voltage changing slowly to the charge and discharge capacitor. A POP noise suppression system is further provided.

Abstract: A method includes receiving at least a first input audio channel and a second input audio channel, and using an inter-channel prediction model to form at least one inter-channel parameter. The first and second input audio channels represent a spatial audio image of an acoustic space. The inter-channel prediction model is a linear prediction model representing a predicted sample of the first input audio channel using a weighted linear combination of samples of the second input audio channel. An apparatus for practicing the method and a corresponding computer program product are also disclosed.

Abstract: An encoding device, a decoding device, and encoding and decoding methods are provided, wherein when a multi-channel signal is encoded with high efficiency, using an adaptive filter, the number of arithmetic operations to update a filter coefficient of the adaptive filter can be reduced. An update range determination unit determines the range of a filter coefficient order (update order range) of a filter coefficient to be updated, among filter coefficients gk(n) of the adaptive filter, on the basis of a mutual correlation function between an input (L) signal and an input (R) signal. The adaptive filter updates the filter coefficient gk(n) of the filter coefficient order (n) to be updated, using a decoding (L) signal and a decoding error (R) signal.

Abstract: A method and system for obtaining an audio signal. In one embodiment, the method comprises receiving a first sound signal at a first microphone arranged at a first height vertically above a substantially flat surface; receiving a second sound signal at a second microphone arranged at a second height vertically above the substantially flat surface; processing a signal provided by the first microphone using a low pass filter; processing a signal provided by the second microphone using a high pass filter; adding the signals processed by the low pass filter and the high pass filter to form a sum signal; and outputting the sum signal as an audio signal.

Abstract: A microphone includes a microelectromechanical system (MEMS) circuit and an integrated circuit. The MEMS circuit is configured to convert a voice signal into an electrical signal, and the integrated circuit is coupled to the MEMS circuit and is configured to receive the electrical signal. The integrated circuit and the MEMS circuit receive a clock signal from an external host. The clock signal is effective to cause the MEMS circuit and integrated circuit to operate in full system operation mode during a first time period and in a voice activity mode of operation during a second time period. The voice activity mode has a first power consumption and the full system operation mode has a second power consumption. The first power consumption is less than the second power consumption. The integrated circuit is configured to generate an interrupt upon the detection of voice activity, and send the interrupt to the host.

Abstract: First analog signals are received from a first microphone, converted into first digital data and stored in a first buffer. A determination is made as to whether voice activity has occurred when voice activity is determined, a voice activity detect signal is sent to an external processor. The external processor responsively provides an exterior clock signal upon receiving the voice activity detect signal. Second analog signals are received from a second microphone, converted into second digital data and stored in a second buffer. The first digital data in the first buffer is not necessarily synchronized in real time with the second digital data in the second buffer. The first digital data from the first buffer and the second digital data from the second buffer is decimated using the external clock to provide decimated output data, the decimated output data having the first digital data and the second digital data aligned in real time.

Abstract: A method for automatically selecting thematically representative music is disclosed. A processor is used for using a theme-related keyword to search a keyword-indexed video repository to retrieve videos associated with the theme-related keyword; analyzing the retrieved videos to select videos with music; and extracting music tracks and features from the selected videos. The method further includes selecting representative music related to the theme from the extracted music tracks using the extracted features; and storing the selected representative music in a processor accessible memory.

Abstract: An active noise reduction earphone. The earphone includes structure for positioning and retaining the earphone in the ear of a user without a headband, active noise reduction circuitry including an acoustic driver with a nominal diameter greater than 10 mm oriented so that a line parallel to, or coincident with, an axis of the acoustic driver and that intersects a centerline of the nozzle intersects the centerline of the nozzle at angle ?>±30 degrees. A microphone is positioned adjacent an edge of the acoustic driver. The earphone is configured so that a portion of the acoustic driver is within the concha of a user and another portion of the acoustic driver is outside the concha of the user when the earphone is in position. An opening coupling the nozzle to the environment includes impedance providing structure in the opening.

Abstract: An audio processing apparatus includes a first microphone, a second microphone, and a masking unit configured to mask movement of air from outside of the apparatus to the second microphone. A filter coefficient is estimated and learned so as to minimize the difference between the output signal of the first microphone and the output signal of the second microphone, thereby suppressing a reverberation component generated in the closed space between the masking unit and the second microphone out of the output signal of the second microphone.

Abstract: A gaming device, during play of a particular game, receives a plurality of audio channels carrying game sounds and determines whether a first of the game sounds overwhelms a second of the game sounds. In response to the determination, the gaming device adjusts a level of one or more of the audio channels such that perceptibility of the second game sound is improved relative to perceptibility of the first game sound prior to the adjustment, wherein for the adjustment, a level of one or more of the audio channels carrying the first game sound is decreased while a level of one or more of the audio channels carrying the second game sound is maintained or increased. The audio channels include three or more audio channels and the adjustment of the level of the audio channels is performed while the three or more audio channels are combined into two stereo channels.

Abstract: An electronic device for converting a multi-channel audio file to a dual channel audio file and vice versa. The multichannel audio file includes a right channel group and a left channel group of channel signals. The electronic device respectively mixes the channel signals of the right channel group and the left channel group according to a mixed matrix to form N mixed signals, and cross embeds the N mixed signals to from a left channel audio signal and a right channel audio signal to compose the dual channel audio file. The electronic device samples, recombines and decodes the left channel audio signal and the right channel audio signal according to a decoding matrix, which is the inverse of the mixed matrix, to revert to the original multi-channel audio file.