Abstract: The present disclosure includes processing a signal to generate a first sub-set of data, transmitting the first sub-set of data for generation of a reconstructed audio signal, the reconstructed audio signal having a fidelity relative to the signal, processing the signal to generate a second sub-set of data and a third sub-set of data, the second sub-set of data defining a second portion of the signal and comprising data that is different than data of the first sub-set of data, and the third sub-set of data defining a third portion of the signal and comprising data that is different than data of the first and second sub-sets of data, comparing a priority of the second sub-set of data to a priority of the third sub-set of data, and transmitting one of the second sub-set of data and the third sub-set of data over the network for improving the fidelity.

Abstract: A multimedia control system which may be expanded without creating or modifying source code. The system is based on a master programmable multimedia controller. The master programmable multimedia controller has a list of unique identifiers that indicate valid subordinate programmable multimedia controllers that are eligible to become part of the system. One or more subordinate programmable multimedia controllers having a matching unique identifier to a unique identifier on the master's list may be added to the system to provide additional control ports, increase audio or video processing or switching capacity. A method is provided for determining whether a particular subordinate programmable multimedia controller may be added to the system.

Abstract: In one aspect, audio streams are added to a mix until the mix is either complete (i.e., all audio streams have been added) or the mix is closed early (i.e., before the mix is complete). In another aspect, audio and video streams are synchronized by playing back the audio stream and then synchronizing display of the video frames to the playback of the audio stream.

Abstract: Technology for smart line-in processing in an audio environment is disclosed. Particularly, the embodiments described herein provide automated source switching in an audio environment where a number of audio sources may exist and volume control.

Abstract: In general, techniques of controlling a plurality of multimedia players in groups are disclosed. According to one aspect of the present invention, a mechanism 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.

Abstract: The present publication describes a data transfer method and system in a digital sound reproduction system. The method comprises method steps for generating a digital audio stream for multiple channels in a host data source, e.g. a computer, the audio stream is formed by multiple consecutive samples, receiving the digital audio stream sent by the host data source through a digital data transmission network by several digital receivers each of which including a microcontroller with a clock, the receivers further including means for generating an audio signal.

Abstract: A method and apparatus are provided for gathering research data indicating exposure to audio content to a user of an audio source device and an audio receiver device. The audio source device wirelessly transmits an audio signal containing the audio content. The audio receiver device receives an audio signal containing the audio content. The audio receiver device includes a speaker to output the audio content to be heard by the user. The apparatus includes a bridge device for wirelessly bridging the audio signal from the audio source device to the audio receiver device. The bridge device includes a speaker for outputting the audio content from the audio signal. The apparatus also includes a monitoring device acoustically coupled to the bridge device for monitoring the audio content output by the speaker of the bridge device, and generating research data identifying the audio content.

Abstract: A method for providing a user interface modified in accordance to audio data, a module including an audio feature extractor and a UI modifier, an apparatus and a computer readable medium. First, audio data is received. Thereafter, an audio feature is extracted from the received audio data. In accordance to the extracted audio feature, the appearance of one of a number of user interface components is modified, e.g. by superposing music visualization effects.

Abstract: Certain embodiments provide methods and systems for managing a sound profile. An example method includes accessing a playback device on a network. The example method includes displaying a graphic interface to allow a user to adjust the sound profile, wherein the sound profile includes a plurality of parameters for user adjustment. The example method includes saving the sound profile. The example method includes processing an audio signal at the playback device according to the sound profile, wherein user adjustments to the sound profile are used to configure the playback device to process the audio signal upon saving the sound profile.

Abstract: In an active muffler having improved response characteristics, a speaker section includes a diaphragm adapted to generate sound, a voice coil for driving the diaphragm, and a distance sensor to detect the movement of the diaphragm. A light generated by the LED is reflected by the diaphragm, the reflected light is detected by a phototransistor to thereby measure the distance to the diaphragm, so that the movement of the diaphragm is detected. Noise is detected by a microphone, and a signal having opposite phase to that of the noise is generated by an opposite-phase generating section. The difference between the opposite-phase signal and the signal of the distance to the speaker from the distance sensor is calculated and inputted to a PID control section. Such a difference indicates the delay of the speaker movement. Feedback control is performed in a direction in which the difference is canceled out.

Abstract: A method for generating an output sequence of samples in response to a first and a second subsequence of samples, the method comprising —applying a weighted overlap-add procedure to the first and second subsequences so as to generate the output sequence of samples, —optimizing a weighting function involved in the weighted overlap-add procedure in response to a measure of matching between the output sequence of samples and one or more target sequences of samples.

Abstract: Aspects of a method and/or system for processing high quality audio in a hardware audio CODEC for audio transmission are provided. In this regard, an audio signal may be down-sampled via a cascaded plurality of filters and sample rate converters in the hardware audio CODEC. Additionally, a portion of each sample of the audio signal may be selected based on an origin of the audio signal. The selected portion of each sample of the audio signal may comprise 16 or 18 bits. The selected portion may be determined based on a type, a class, a manufacturer identifier, and/or a model identifier of the origin the audio signal. Coefficients of the filters may be configured based on the origin of the audio signal. One or more of the filters may comprise one or more cascaded biquads. The sample rate converters may comprise one or more CIC decimation filters.

Abstract: Method and devices for processing audio signals based on sound profiles are provided. A reproduction device can request a sound profile based on user information, device information, media metadata or a combination. The sound profiles can be customized and shared across multiple devices. User interfaces allow for the input of information that allows the reproduction device or a server in the cloud to select, modify, store, and analyze sound profiles. Deeper analysis allows for the improvement of sound profiles for individuals and groups. Intensity scoring of a music library can also be conducted.

Abstract: A sound enhancement system is configured such that sound emitting and collecting apparatuses 100 to 105 are connected to each other via a transmission line LT. The sound emitting and collecting apparatuses 100 to 105 are grouped into a master and a slave. The master sound emitting and collecting apparatus 100 generates a reference clock signal to be supplied to slave sound emitting and collecting apparatuses 101 to 105. The slave sound emitting and collecting apparatuses 101 to 105 form a sound collection beam signal in synchronization with the reference clock signal, and then output the resulting signal to the master sound emitting and collecting apparatus 100 without performing acoustic coupling suppressing process. The master sound emitting and collecting apparatus 100 collectively executes the acoustic coupling suppressing process to each sound collection beam signal to produce a sound emitting signal.

Abstract: In general, techniques of controlling a plurality of multimedia players in groups are disclosed. According to one aspect of the present invention, a mechanism 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.

Abstract: Integrated processing of multimedia signals can eliminate unnecessary signal processors and converters without losing the functionality of typical home entertainment system components. The integrated multimedia system includes a main player that captures and processes signals digitally. The main player may adjust the audio signal to provide audio output of equal loudness across all frequencies by accounting for sensitivity of the human ear for sounds of varying frequencies. The main player can also account for perceived differences in loudness based on the angle of a listener to a speaker by detecting the position of a user and making an adjustment accordingly. The invention further provides a speaker that has embedded performance characteristics or an identifier that allows the system to provide an optimal speaker driving current for a particular system or determine how that speaker would be best implemented in the integrated system.

Abstract: A gain control module includes an amplifier, a least significant bit (LSB) gain stage, and a most significant bit (MSB) gain stage. The amplifier includes a first input, a second input, and an output. The LSB gain stage produces a LSB gain based on an LSB portion of a gain control signal, wherein the LSB gain stage receives an input signal. The MSB gain stage produces an MSB gain based on an MSB portion of the gain control signal, wherein the MSB gain stage is coupled to the LSB gain stage, the first input of the amplifier, and the output of the amplifier, wherein the gain control module amplifies the input signal in accordance with the gain control signal.

Abstract: The present invention is to provide a method of efficiently detecting music scenes from television broadcasts. A music-segment detection method according to the present invention includes a data input step of inputting data including an audio and video signals; a plurality of candidate-music-segment detection steps of detecting candidate music-segments from the audio signal based on criteria different from each other; and a music-segment detection step of detecting a music segment based on the plurality of detected candidate music-segments. Thereby, high detection accuracy can be ensured for a wide variety of music with various tones, such as rock, popular ballad, and classic.

Abstract: An audio dynamics processor or processing method that uses a reset mechanism or process in order to adapt quickly to content changes in the audio signal. A reset signal may be generated by analyzing the audio signal itself or the reset may be triggered from an external event such as a channel change on a television set or an input selection change on an audio/visual receiver. In the case of an external trigger, one or more indicators of the state of the dynamics processor for a current audio source may be saved and associated with that audio source before switching to a new audio source. Then, if the system switches back to the first audio source, the dynamics processor may be reset to the state previously stored or an approximation thereof.

Abstract: In general, techniques of controlling a plurality of multimedia players in groups are disclosed. According to one aspect of the present invention, a mechanism 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.