Sony Corporation

Sony Pictures Entertainment, Inc.

Dts, Inc.

Srs Labs, Inc.

Other

Sony Electronics Inc

Richard J Oliver

Roger Mather Duvall

Jeffrey Mark Claar

Jun Yang

Alan D Kraemer

Robert C Maling, Iii

Sarah Yang

James Tracey

Casper W Barnes

Thomas C K Yuen

Jean-Marc Jot

Paul M Embree

Aaron Kube Warner

Xing He

Jeff Claar

Robert C Iii Maling

A system can be provided for increasing loudness of an audio signal to present a perceived loudness to a listener that is greater than a loudness provided natively by a loudspeaker. The system can include one or more of the following: a frequency suppressor, a loudness adjuster, an equalizer, and a distortion control module. The frequency suppressor can increase headroom in the audio signal by filtering out low and/or high frequencies. The loudness adjuster can calculate a loudness of the audio signal and apply a gain to the audio signal to increase the loudness. The equalizer can further increase headroom by attenuating portions of a passband of the loudspeaker's frequency response. The distortion control module can induce partial harmonic distortion in the audio signal to further increase loudness.

System for increasing perceived loudness of speakers

Systems and methods to provide an audio spectral correction can be implemented in a variety of applications. Such systems and methods can include an equalizer design that uses efficient filter sections tuned with a closed form algorithm to give an accurate and intuitive frequency response with low complexity and minimal processing overhead. Embodiments can enable efficient and accurate control of reverberation time vs. frequency in artificial reverberators. Embodiments can enable efficient realization of a cascade combination of multi-band equalization functions as a single multi-band equalizer function, in which such functions can be expressed as graphic equalization functions using the same set of center frequencies as the cascade combination of multi-band equalization functions. Additional apparatus, systems, and methods are disclosed.

Spectral correction of audio signals

Various embodiments of systems and methods for reducing audio noise are disclosed. One or more sound components such as noise and network tone can be detected based on power spectrum obtained from a time-domain signal. Results of such detection can be used to make decisions in determination of an adjustment spectrum that can be applied to the power spectrum. The adjusted spectrum can be transformed back into a time-domain signal that substantially removes undesirable noise(s) and/or accounts for known sound components such as the network tone.

Systems and methods for reducing audio noise

In certain embodiments, an improved audio equalization filter can be generated by frequency warping one or more digital filters having a plurality of frequency bands. Frequency warping can include, for example, transforming at least some of the frequency bands of the one or more digital filters into lower frequency bands. As a result, in various implementations the audio equalization filter may be more accurate than certain currently-available IIR equalization filters. The audio equalization filter may also be more computing-resource efficient than certain currently-available FIR equalization filters.

Frequency-warped audio equalizer

An adaptive audio system can be implemented in a communication device. The adaptive audio system can enhance voice in an audio signal received by the communication device to increase intelligibility of the voice. The audio system can adapt the audio enhancement based at least in part on levels of environmental content, such as noise, that are received by the communication device. For higher levels of environmental content, for example, the audio system might apply the audio enhancement more aggressively. Additionally, the adaptive audio system can detect substantially periodic content in the environmental content. The adaptive audio system can further adapt the audio enhancement responsive to the environmental content.

System for processing an audio signal to enhance speech intelligibility

An acoustic correction apparatus processes a pair of left and right input signals to compensate for spatial distortion as a function of frequency when said input signals are reproduced through loudspeakers in a sound system. The sound-energy of the left and right input signals is separated and corrected in a first low-frequency range and a second high-frequency range. The resultant signals are recombined to create image-corrected audio signals having a desired sound-pressure response when reproduced by the loudspeakers in the sound system. The desired sound-pressure response creates an apparent sound image location with respect to a listener. The image-corrected signals can also be spatially-enhanced to broaden the apparent sound image and improve the low frequency characteristics of the sound when played on small loudspeakers.

Acoustic correction apparatus

Reproduction of stereophonic audio information over a single speaker requires summing multiple stereo channels. When signals having approximately equal magnitudes and approximately opposite phases at a frequency are added together, the audio information at the frequency is lost. To preserve areas of potential cancellation and potential audio information loss, the audio enhancement system adjusts the phase relationship between the stereophonic channels. To avoid the loss of the spatial content of the stereo signal, the audio enhancement system determines the difference information that exists between different stereophonic channels. The audio enhancement system enhances the difference information and mixes the enhanced difference information with the phase adjusted signals to generate an enhanced monophonic output.

Systems and methods of spatial image enhancement of a sound source

A method and system for providing enhanced editing capability of media data. Media data arriving in a continuous stream is saved in a buffer. When a user enters record mode, some data received before record mode was entered is designated for permanent storage. Similarly, all data received while in record mode is designated for permanent storage. Once the user leaves record mode, additional data received after leaving record mode is also designated for permanent storage. In due course all data designated for permanent storage is stored to a mass storage device. Any data in the buffer not designated for permanent storage will be overwritten without being stored to the mass storage device. In this manner, a media file having record handles is created. The record handles permit a user to slide the nominal entry and exit from record mode to accommodate timing and synchronization errors.

Method and system for providing enhanced editing capabilities

Late audio buffer processing for intelligently muting audio output when audio buffers are not read fast enough from disk into digital signal processing (DSP) memory. Given a playback audio recorder, each channel of audio is associated with a forward and a reverse late threshold signals which are enabled when the processing of an audio buffer for that channel is delayed. When an audio buffer for that channel is delayed, the late threshold signal sets in motion the muting of that specific audio channel for a predetermined amount of time. If and when the system catches up, normal playback resumes. Otherwise, the channel stays muted.

Method and apparatus for late buffer processing

A method and apparatus for storing digitized audio/video tracks. A plurality of lists are created wherein each of the plurality of lists defines a characteristic of the audio/visual data. The plurality of lists are linked according to a data structure such that performing an edit operation on a portion of the audio/visual data results in the edit operation being applied to the plurality of lists.

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Capistrano Beach, CA, United States of America

Richard J Oliver