Abstract: According to various embodiments, a loudspeaker horn and cabinet are designed to achieve a sound coverage pattern characterized by narrow vertical dispersion and a wide horizontal dispersion. A loudspeaker horn may comprise at least two horn sections, each extending from an inlet to a mouth. A first plurality of outlet channels is disposed in an interleaved column with a second plurality of outlet channels. A loudspeaker cabinet may comprise a primary enclosure having a front wall, the front wall having an aperture in which a low frequency loudspeaker driver is mounted. The loudspeaker cabinet further comprises a top baffle section having a top end and a bottom baffle section having a bottom end, each extending vertically from the primary enclosure. The top baffle section has a first width that gradually increases towards the top end and the bottom section has a second width that gradually increases towards the bottom end.

Abstract: Systems and methods for leveling loudness variation in an audio signal are described. Embodiments use both a perceptual leveling algorithm and a standards-based loudness measure together to minimize audio process artifacts and ensure that the measured loudness of the processed audio is close to a required measure, according to a particular standard measurement of loudness. These systems and methods can be used either offline or in real-time.

Abstract: Scaling, by a desired amount sm, the overall perceived loudness Lm of a multichannel audio signal, wherein perceived loudness is a nonlinear function of signal power P, by scaling the perceived loudness of each individual channel Lc by an amount substantially equal to the desired amount of scaling of the overall perceived loudness of all channels sm, subject to accuracy in calculations and the desired accuracy of the overall perceived loudness scaling sm. The perceived loudness of each individual channel may be scaled by changing the gain of each individual channel, wherein gain is a scaling of a channel's power. Optionally, in addition, the loudness scaling applied to each channel may be modified so as to reduce the difference between the actual overall loudness scaling and the desired amount of overall loudness scaling.

Abstract: The invention relates to modifying the loudness of an audio signal by measuring the weighted broadband level of the audio signal and modifying that weighted broadband level as a function of a spectral localization estimate of the audio signal.

Abstract: A spread spectrum data hiding for audio signals is described. A set of pseudo-random noise sequences is added to an audio signal according to a data to be embedded. A masking curve is used to shape the added noise. A transient detection step can be used to control whether a shaped noise sequence is to be added or not. Embedded information is detected by first performing a whitening step and then performing a phase-only correlation with a same set of pseudo-random noise sequences. A detection method that is based on correlation of multiplexed noise sequences with a noise sequence embedded in the audio is also described.

Abstract: Scaling, by a desired amount sm, the overall perceived loudness Lm of a multichannel audio signal, wherein perceived loudness is a nonlinear function of signal power P, by scaling the perceived loudness of each individual channel Lc by an amount substantially equal to the desired amount of scaling of the overall perceived loudness of all channels sm, subject to accuracy in calculations and the desired accuracy of the overall perceived loudness scaling sm. The perceived loudness of each individual channel may be scaled by changing the gain of each individual channel, wherein gain is a scaling of a channel's power. Optionally, in addition, the loudness scaling applied to each channel may be modified so as to reduce the difference between the actual overall loudness scaling and the desired amount of overall loudness scaling.

Abstract: Techniques for adaptive processing of media data based on separate data specifying a state of the media data are provided. A device in a media processing chain may determine whether a type of media processing has already been performed on an input version of media data. If so, the device may adapt its processing of the media data to disable performing the type of media processing. If not, the device performs the type of media processing. The device may create a state of the media data specifying the type of media processing. The device may communicate the state of the media data and an output version of the media data to a recipient device in the media processing chain, for the purpose of supporting the recipient device's adaptive processing of the media data.

Abstract: A method for controlling the loudness of auditory events in an audio signal. In an embodiment, the method includes weighting the auditory events (an auditory event having a spectrum and a loudness), using skewness in the spectra and controlling loudness of the auditory events, using the weights. Various embodiments of the invention are as follows: The weighting being proportionate to the measure of skewness in the spectra; the measure of skewness is a measure of smoothed skewness; the weighting is insensitive to amplitude of the audio signal; the weighting is insensitive to power; the weighting is insensitive to loudness; and any relationship between signal measure and absolute reproduction level is not known at the time of weighting; the weighting includes weighting auditory-event-boundary importance, using skewness in the spectra.

Abstract: The invention relates to modifying the loudness of an audio signal by measuring the weighted broadband level of the audio signal and modifying that weighted broadband level as a function of a spectral localization estimate of the audio signal.

Abstract: A method for controlling the loudness of auditory events in an audio signal. In an embodiment, the method includes weighting the auditory events (an auditory event having a spectrum and a loudness), using skewness in the spectra and controlling loudness of the auditory events, using the weights. Various embodiments of the invention are as follows: The weighting being proportionate to the measure of skewness in the spectra; the measure of skewness is a measure of smoothed skewness; the weighting is insensitive to amplitude of the audio signal; the weighting is insensitive to power; the weighting is insensitive to loudness; and any relationship between signal measure and absolute reproduction level is not known at the time of weighting; the weighting includes weighting auditory-event-boundary importance, using skewness in the spectra.

Abstract: One or a combination of two or more specific loudness model functions selected from a group of two or more of such functions are employed in calculating the perceptual loudness of an audio signal. The function or functions may be selected, for example, by a measure of the degree to which the audio signal is narrowband or wideband. Alternatively or with such a selection from a group of functions, a gain value G[t] is calculated, which gain, when applied to the audio signal, results in a perceived loudness substantially the same as a reference loudness. The gain calculating employs an iterative processing loop that includes the perceptual loudness calculation.

Abstract: A process for combining audio channels combines the audio channels to produce a combined audio channel and dynamically applies one or more of time, phase, and amplitude or power adjustments to the channels, to the combined channel, or to both the channels and the combined channel. One or more of the adjustments are controlled at least in part by a measure of auditory events in one or more of the channels and/or the combined channel. Applications include the presentation of multichannel audio in cinemas and vehicles. Not only methods, but also corresponding computer program implementations and apparatus implementations are included.

Abstract: A coded signal conveys encoded audio information and metadata that may be used to control the loudness and dynamic range of the audio information during its playback. If the values for these metadata parameters are set incorrectly, annoying fluctuations in loudness during playback can result. The present invention overcomes this problem by detecting incorrect metadata parameter values in the signal and replacing the incorrect values with corrected values.

Abstract: An encoder and associated vector estimation method and system (1) for processing a sequence of input vectors (y0 to yT) each comprising a plurality of elements. The vector estimation system (1) has a digital filter (2) with a filter vector input (3) for receiving said sequence of input vectors (y0 to yT) and a predictor gain input (4) for controlling characteristics of the filter (2). The filter (2) is a Kalman filter and has both a current slowly evolving filter estimate output (6) and a previous slowly evolving filter estimate output (20). The current slowly evolving filter estimate output (6) provides a current filtered estimate value of a slowly evolving component of said sequence of input vectors (y0 to yT) and the previous slowly evolving filter estimate output (20) provides a previous filtered estimate value of the slowly evolving component of said sequence of input vectors (y0 to yT).

Abstract: Estimates of spectral magnitude and phase are obtained by an estimation process using spectral information from analysis filter banks such as the Modified Discrete Cosine Transform. The estimation process may be implemented by convolution-like operations with impulse responses. Portions of the impulse responses may be selected for use in the convolution-like operations to trade off between computational complexity and estimation accuracy. Mathematical derivations of analytical expressions for filter structures and impulse responses are disclosed.

Abstract: Methods for splicing PCM audio frames form modified frames by appending to each frame either a portion of the next preceding frame or the next following frame. According to a first approach, splices are obtained by fading up and fading down a frame end and a frame appendage only at a splice point, and overlapping and combining to provide a crossfade at the splice. Alternatively, every frame end and frame appendage is faded up and faded down, overlapped and combined. A subtractive method of providing complementary fade-up and fade-down reduces rounding errors.

Abstract: An encoder and associated vector estimation method and system (1) for processing a sequence of input vectors (y0 to yT) each comprising a plurality of elements. The vector estimation system (1) has a digital filter (2) with a filter vector input (3) for receiving said sequence of input vectors (y0 to yT) and a predictor gain input (4) for controlling characteristics of the filter (2). The filter (2) is a Kalman filter and has both a current slowly evolving filter estimate output (6) and a previous slowly evolving filter estimate output (20). The current slowly evolving filter estimate output (6) provides a current filtered estimate value of a slowly evolving component of said sequence of input vectors (y0 to yT) and the previous slowly evolving filter estimate output (20) provides a previous filtered estimate value of the slowly evolving component of said sequence of input vectors (y0 to yT).