Abstract: An audio signal attenuation system and method for detecting an audio emergency warning signal (or alarm) in a vehicle in which an audio signal is being played. Embodiments of the system and method make it easier for a police, fire, or other emergency alarm or siren to be heard in a loud or noisy listening environment when audio signal is being reproduced. This is achieved using selective frequency attenuation, which identifies a frequency of the alarm and then selectively attenuates the alarm frequency in the audio signal. Moreover, direction data that includes information about from which direction the alarm is coming can be used to selectively attenuate the alarm frequency in certain channels (or speakers) of the audio signal. In some embodiments, audio cues are used to alert the listener to the alarm signal and are adjusted based on alarm distance from the vehicle, speed, and the type of alarm.

Abstract: An input audio signal is equalized to form an output audio signal on the basis of an intended listening sound pressure level, the output capabilities of a particular playback device, and unique hearing characteristics of a listener. An intended listening level is first determined based on the properties of the audio signal and a mastering sound level. The intended listening level is used to determine an optimal sound pressure level for the particular playback device based on its capabilities and any master volume gain. These two levels are used to determine how much louder to make individual frequencies based on data pertaining to human auditory perception, either standardized or directly measured. The audio is further compensated on the basis of hearing loss data, again either standardized or directly measured, after optionally extending the signal bandwidth. The final, compensated audio signal is sent to the playback device for playback.

Abstract: A system and method for estimating a reverberation time is provided. The method includes estimating at least one room response of an audio capture environment with an acoustic echo canceller and generating an estimate of the reverberation time of the audio capture environment based on the at least one room response from the acoustic echo canceller.

Abstract: A predictive pattern high-frequency reconstruction system and method that finds patterns in high-frequency components of an audio signal, encodes the audio signal into an encoded bitstream along with pattern information, and then uses the patterns to reconstruct the high-frequency components during decoding. The high-frequency components can be reconstructed using the pattern information alone. Embodiments of the system and method map normalized subband signals of the audio signal to a scaled representation of a time-frequency grid containing multiple tiles and perform statistical analysis on each tile to estimate subband parameters and determine whether a pattern exists. If a pattern does exist, it can be encoded in the encoded bitstream, transmitted, and used to reconstruct the high-frequency components at the decoder. A direct search technique and a fast Fourier transform (FFT) technique may be used to perform the statistical analysis.

Abstract: A constant-power pairwise panning upmixing system and method for upmixing from a two-channel stereo signal to a multi-channel surround sound (having more than two channels). Each output channel is some combination of the two input channels. Closed-form solutions are used to calculate dematrixing coefficients that are used to weight each input channel. The dematrixing coefficients are computed based on an inter-channel level difference and an inter-channel phase difference between the two input signals. The weighted input channels then are mixed uniquely for each output channel to generate a surround sound output from the stereo input signal. Each dematrixing coefficient has an in-phase component and an out-of-phase component. The phase coefficients for each component vary in time and are based on the phase difference between the input signals. The resultant surround sound output faithfully simulates the audio content as originally mixed.

Abstract: An audio depth dynamic range enhancement system and method for enhancing the dynamic range of depth in audio sound systems as perceived by a human listener. Embodiments of the system and method process an input audio signal by applying a gain function to at least one of a plurality of sub-signals of the audio signal having different values of a spatial depth parameter. The sub-signals are combined to produce a reconstructed audio signal carrying modified audio information. The reconstructed audio signal is output from the system and method for reproduction by the audio sound system. The gain function alters the gain of the at least one of the plurality of sub-signals such that the reconstructed audio signal, when reproduced by the audio sound system, results in modified depth dynamic range of the audio sound system with respect to the spatial depth parameter.

Abstract: A post-encoding bitrate reduction system and method for generating one more scaled compressed bitstreams from a single encoded plenary file. The plenary file contains multiple audio object files that were encoded separately using a scalable encoding process having fine-grained scalability. Activity in the data frames of the encoded audio object files at a time period are compared with each other to obtain a data frame activity comparison. Bits from an available bitpool are assigned to all of the data frames based on the data frame activity comparison and corresponding hierarchical metadata. The plenary file is scaled down by truncating bits in the data frames to conform to the bit allocation. In some embodiments frame activity is compared to a silence threshold and the data frame contains silence if the frame activity is less than or equal to the threshold and minimal bits are used to represent the silent frame.

Abstract: Various exemplary embodiments relate to a method and apparatus for processing audio signals to influence the reproduction of the audio signals. The apparatus may include a speaker, a headphone (over-the-ear, on-ear, or in-ear), a microphone, a computer, a mobile device, a home theater receiver, a television, a Blu-ray (BD) player, a compact disc (CD) player, a digital media player, or the like. The apparatus may be configured to receive a virtualization profile including a digital audio filter with a design sample rate, resample the virtualization profile to a different sample rate, filter the audio signal with the resampled virtualization profile, and reproduce the filtered audio signal as sound.

Abstract: Various embodiments relate to a system and method for decorrelating an audio signal with a hybrid filter. The hybrid filter is generated by first generating a decorrelation filter. A frequency-dependent warping is applied to the decorrelation filter. The warped decorrelation filter is then mixed with a carrier filter to generate the hybrid filter. The carrier filter may include filters for spatial processing of an audio signal, filters for upmixing an audio signal, and/or filters for downmixing an audio signal.

Abstract: There is disclosed methods and apparatus for decomposing a signal having a plurality of channels into direct and diffuse components. The correlation coefficient between each pair of signals from the plurality of signals may be estimated. A linear system of equations relating the estimated correlation coefficients and direct energy fractions of each of the plurality of channels may be constructed. The linear system may be solved to estimate the direct energy fractions. A direct component output signal and a diffuse component output signal may be generated based in part on the direct energy fractions.

Abstract: A method and apparatus processes multi-channel audio by encoding, transmitting or recording “dry” audio tracks or “stems” in synchronous relationship with time-variable metadata controlled by a content producer and representing a desired degree and quality of diffusion. Audio tracks are compressed and transmitted in connection with synchronized metadata representing diffusion and preferably also mix and delay parameters. The separation of audio stems from diffusion metadata facilitates the customization of playback at the receiver, taking into account the characteristics of local playback environment.

Abstract: Lossy compression and transmission of a downmixed composite signal having multiple tracks and objects, including a downmixed signal, is accomplished in a manner that reduces the bit-rate requirement as compared to redundant transmission or lossless compression, while reducing upmix artifacts. A compressed residual signal is generated and transmitted along with a compressed total mix and at least one compressed audio objects. In the reception and upmix aspect the invention decompresses a downmixed signal and other compressed objects, calculates an approximate upmix signal, and corrects specific base signals derived from the upmix, by subtracting a decompressed residual signal. The invention thus allows lossy compression to be used in combination with downmixed audio signals for transmission through a communication channel (or for storage).

Abstract: A post-encoding bitrate reduction system and method for generating one more scaled compressed bitstreams from a single encoded plenary file. The plenary file contains multiple audio object files that were encoded separately using a scalable encoding process having fine-grained scalability. Activity in the data frames of the encoded audio object files at a time period are compared with each other to obtain a data frame activity comparison. Bits from an available bitpool are assigned to all of the data frames based on the data frame activity comparison and corresponding hierarchical metadata. The plenary file is scaled down by truncating bits in the data frames to conform to the bit allocation. In some embodiments frame activity is compared to a silence threshold and the data frame contains silence if the frame activity is less than or equal to the threshold and minimal bits are used to represent the silent frame.

Abstract: Devices and methods are adapted to characterize a multi-channel loudspeaker configuration, to correct loudspeaker/room delay, gain and frequency response or to configure sub-band domain correction filters.

Abstract: A multiplet-based spatial matrixing codec and method for reducing channel counts (and thus bitrates) of high-channel count (seven or more channels) multichannel audio, optimizing audio quality by enabling tradeoffs between spatial accuracy and basic audio quality, and converting audio signal formats to playback environment configurations. An initial N channel count is reduced to M channels by spatial matrix mixing to a lower number of channels using multiplet pan laws. The multiplet pan laws include doublet, triplet, and quadruplet pan laws. For example, using a quadruplet pan law one of the N channels can be downmixed to four of the M channels to create a quadruplet channel. Spatial information as well and audio content is contained in the multiplet channels. During upmixing the downmixed channel is extracted from the multiplet channels using the corresponding multiplet pan law. The extracted channel then is rendered at any location within a playback environment.

Abstract: A method and apparatus processes multi-channel audio by encoding, transmitting or recording “dry” audio tracks or “stems” in synchronous relationship with time-variable metadata controlled by a content producer and representing a desired degree and quality of diffusion. Audio tracks are compressed and transmitted in connection with synchronized metadata representing diffusion and preferably also mix and delay parameters. The separation of audio stems from diffusion metadata facilitates the customization of playback at the receiver, taking into account the characteristics of local playback environment.

Abstract: Devices and methods are adapted to characterize a multi-channel loudspeaker configuration, to correct loudspeaker/room delay, gain and frequency response or to configure sub-band domain correction filters.

Abstract: A method and apparatus processes multi-channel audio by encoding, transmitting or recording “dry” audio tracks or “stems” in synchronous relationship with time-variable metadata controlled by a content producer and representing a desired degree and quality of diffusion. Audio tracks are compressed and transmitted in connection with synchronized metadata representing diffusion and preferably also mix and delay parameters. The separation of audio stems from diffusion metadata facilitates the customization of playback at the receiver, taking into account the characteristics of local playback environment.

Abstract: The present invention provides a novel end-to-end solution for creating, encoding, transmitting, decoding and reproducing spatial audio soundtracks. The provided soundtrack encoding format is compatible with legacy surround-sound encoding formats, so that soundtracks encoded in the new format may be decoded and reproduced on legacy playback equipment with no loss of quality compared to legacy formats.

Abstract: There are provided methods and an apparatus for conditioning an audio signal. According to one aspect of the present invention there is included a method for conditioning an audio signal having the steps of: receiving at least one audio signal, each audio signal having at least one channel, each channel being segmented into a plurality of frames over a series of time; calculating at least one measure of dynamic excursion of the audio signal for a plurality of successive segments of time; filtering the audio signal into a plurality of subbands, each frame being represented by at least one subband; deriving a dynamic gain factor from the successive segments of time; analyzing at least one subband of the frame to determine if a transient exists in the frame; and applying the dynamic gain factor to each frame having a transient.