Abstract: Systems and techniques for removing a sound recording from an audio recording (e.g., an audio recording embedded in a media file) are presented. The system can include an identification component, a first subtraction component and a second subtraction component. The identification component identifies a sound recording in a mixed audio recording. The first subtraction component determines a local linear transformation of the sound recording and subtracts the local linear transformation of the sound recording from the mixed audio recording to generate a new mixed audio recording. The second subtraction component compares one or more segments of the sound recording with one or more corresponding segments of the new mixed audio recording and reduces a power level of the new mixed audio recording based at least in part on correlation of the one or more corresponding segments with the one or more segments.

Abstract: A system provides a model for use within a digital environment. The model comprises at least one music segment, and supplies music for the digital environment. The system identifies a location within the digital environment. The location requires at least one music segment supplied by the model. The system selects at least one music segment to play within the digital environment. The music segment is selected based on the model, the location requiring the at least one music segment, and the digital environment. The selecting of the music segment is operable to be modified based on the model, the location requiring the at least one music segment, and the digital environment.

Abstract: This specification describes technologies relating to modifying digital audio data using a meta-parameter. A method is provided that includes receiving digital audio data; receiving an input modifying a meta-parameter to a particular meta-parameter value associated with two or more parameters according to a particular mapping, each of the two or more parameters being associated with the digital audio data; modifying at least a first parameter of the two or more parameters based on the particular meta-parameter value and the particular mapping such that a first meta-parameter value causes a modification of the first parameter associated with a volume control of the digital audio data and a second meta-parameter value causes a modification of a second parameter of the two or more parameters associated with dynamic range control of the digital audio data; and generating modified digital audio data from the two or more modified parameters of the digital audio data.

Abstract: The present invention sets forth a method for supporting enhanced audio on a graphics processing unit (GPU) in a computing device having a graphics subsystem. In one embodiment, the method includes the steps of determining whether an option of a GPU audio output is enabled and the graphics subsystem and a first external output device is connected, and routing a first audio stream to the GPU of the graphics subsystem for processing when the option of the GPU audio output is enabled and the graphics subsystem and the first external output device is in connection and causing the processed first audio stream to be transferred along a first transmission path to the first external output device, or otherwise causing a second audio stream to be transferred along a second transmission path to a second external output device.

Abstract: A device (200) for processing an audio data stream, the device (200) comprising a transient detection unit (201) adapted to detect a transient portion of an audio input data stream (202), and a harmonics generator (203) adapted to generate an audio output data stream (204) based on the audio input data stream (202), the audio output data stream (204) comprising a sequence of harmonics (205) generated only from a non-transient portion of the audio input data stream (202).

Abstract: Technology for grouping, consolidating, and pairing individual playback devices with network capability (players) to stimulate a multi-channel listening environment is disclosed. Particularly, the embodiments described herein enable two or more playback devices to be paired, such that multi-channel audio is achieved. Such embodiments may be used to produce stereo and multi-channel audio environments for television and movies.

Abstract: In general, user interfaces for controlling a plurality of multimedia players in groups are disclosed. According to one aspect of the present invention, a user interface is provided to allow a user to group some of the players according to a theme or scene, where each of the players is located in a zone. When the scene is activated, the players in the scene react in a synchronized manner. For example, the players in the scene are all caused to play a multimedia source or music in a playlist, wherein the multimedia source may be located anywhere on a network. The user interface is further configured to illustrate graphically a size of a group, the larger the group appears relatively, the more plays there are in the group.

Abstract: Embodiments of an electromechanical transducer are presented. The electromechanical transducer includes a plurality of magnets and a substrate having a patterned conductive coil of at least one turn. The plurality of magnets are arranged such that a continuous magnetic flux path passes through each of the plurality of magnets. The substrate is located within an airgap between a given magnet of the plurality of magnets and an adjacent magnet of the plurality of magnets. The continuous magnetic flux path passes through at least a portion of the patterned conductive coil due to the location of the substrate within the airgap.

Abstract: User is allowed to designate a desired mode defining the respective numbers of channels and mixing buses, and processing for mixing input signals of the number of channels corresponding to the designated mode is performed repetitively to generate signals for the individual buses. The time of arrival of the last step in the mixing processing for the number of channels, corresponding to the designated mode, is detected to output an accumulation result obtained at the last step, and new accumulation is started with a digital audio signal inputted at a step following the last step. Digital audio signals processed by a first signal processing circuit are stored into a memory and transmitted to a second signal processing circuit via a cascade-connection. The second signal processing circuit adds the audio signal, processed for each of the steps, to audio signals input via the cascade-connection and writes added signal into the memory.

Abstract: A mobile device that is capable of automatically starting and ending the recording of an audio signal captured by at least one microphone is presented. The mobile device is capable of adjusting a number of parameters related with audio logging based on the context information of the audio input signal.

Abstract: An acoustic playback system including a digital filter; and a plurality of digital modulators each of which output a digital signal to one of a plurality of speakers configured with speakers driven by digital signals having different play back bandwidths; wherein the digital filter converts a digital audio signal which is input into a plurality of digital audio signals of a plurality of frequency bandwidths corresponding to play back bandwidths of the plurality of speakers, and outputs each of the digital audio signals of the plurality of frequency bandwidths to one of the plurality of digital modulators; each of the plurality of digital modulators outputs the modulated digital signal to the speaker of a play back bandwidth corresponding to a frequency bandwidth of the digital audio signal which is input by performing miss match shaping after noise shaping to a digital audio signal which is input; and each number of bits of a digital signal which is output by each of the digital modulators is different is pro

Abstract: A method and system for resynchronizing an embedded multimedia system using bytes consumed in an audio decoder. The bytes consumed provides a mechanism to compensate for bit error handling and correction in a system that does not require re-transmission. The audio decoder keeps track of the bytes consumed and periodically reports the bytes consumed. A host microprocessor indexes the actual bytes consumed since bit errors may have been handled or corrected to a predetermined byte count to determine whether resynchronization is necessary.

Abstract: The present invention relates to an audio signal controller adapted to receiving first and second digital audio signals and estimating a signal feature of the first or second digital audio signal. The estimated signal feature is compared with a predetermined feature criterion and the audio signal controller switches from conveying the first digital audio signal to conveying the second digital audio signal to a controller output, or vice versa, at a zero-crossing of the first digital audio signal or the second digital audio signal based on the comparison between the estimated signal feature and the predetermined feature criterion.

Abstract: Individual adjustment of audio volume and video properties in a computer network conference environment is provided. For audio adjustment, a buffer collects incoming streams; a stream decoder decodes the buffered audio streams, a gain adjustment applies a gain increase or decrease to the individual audio stream, and a mixer combines each of the individual signals together. The gain adjustment module receives input from a user interface control associated with each participant, and adjusts the volume of that participant's stream accordingly. If a requested increase in gain would cause an overflow of the signal, only a gain increase that will avoid such overflow is applied. Video properties such as brightness, contrast and saturation are also adjustable. Properties of the user's transmitted audio and video streams are also adjustable.

Abstract: Certain aspects of a method and system for a flexible multiplexer and mixer (FMM) are disclosed. Aspects of one method may include mixing primary audio information and secondary audio information of sampled received audio data based on corresponding metadata information to generate mixed output audio data. The generated mixed output audio data may be pulled through a plurality of pipeline stages.

Abstract: To provide an audio signal processing apparatus which can perform, with low operation amount, audio signal processing that is either time stretch and/or compression processing or frequency modulation processing. The audio signal processing apparatus is intended to transform an input audio signal sequence using a predetermined adjustment factor. The audio signal processing apparatus includes a filter bank (2601) which transforms the input audio signal sequence into Quadrature Mirror Filter (QMF) coefficients using a filter for Quadrature Mirror Filter analysis (a QMF analysis filter) and an adjusting unit (2602) which adjusts the QMF coefficients based on a predetermined adjustment factor.

Abstract: Recording devices are often used to record discussions, conferences, meetings and the like. The recorded information is reviewed or processed afterwards for example in order to prepare minutes or notes. The known state of the art proposes to use devices comprising two receptacles for tapes. But this solution adds cost to the recording devices, which will not be accepted from all potential buyers as only a small number of the buyers will actually use this functionality.

Abstract: Technology for grouping, consolidating, and pairing individual playback devices with network capability (players) to stimulate a multi-channel listening environment is disclosed. Particularly, the embodiments described herein enable two or more playback devices to be paired, such that multi-channel audio is achieved. Such embodiments may be used to produce stereo and multi-channel audio environments for television and movies.

Abstract: A system that incorporates teachings of the present disclosure may include, for example, a set top box having a controller to receive a request for reverberation in an environment having a plurality of media devices, and adjust a time delay for audio signals presented by one of the plurality of media devices operably coupled to the set top box based at least in part on the request. Other embodiments are disclosed.

Abstract: Method and devices for testing a headphone with increased sensation are provided. The headphone can filter and amplify low frequency audio signals, which are then sent to a haptic device in the headphone. The haptic device can cause bass sensations at the top of the skull and at both ear cups. The testing system can evaluate the haptic and acoustic sensations produced by the headphone to evaluate if they have been properly assembled and calibrate the headphones if necessary.