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: A multi-rate audio processing system and method provides real-time measurement and processing of amplitude/phase changes in the transition band of the lowest frequency subband caused by the audio processing that can be used to apply amplitude/phase compensation to the higher subband(s). Tone signals may be injected into the transition band to provide strong tonal content for measurement and processing. The real-time measurement and compensation adapts to time-varying amplitude/phase changes regardless of the source of the change (e.g. non-linear time-varying linear or user control parameters) and provides universal applicability for any linear audio processing.

Abstract: Loudness control systems or methods may normalize audio signals to a predetermined loudness level. If the audio signal includes moderate background noise, then the background noise may also be normalized to the target loudness level. Noise signals may be detected using content-versus-noise classification, and a loudness control system or method may be adjusted based on the detection of noise. Noise signals may be detected by signal analysis in the frequency domain or in the time domain. Loudness control systems may also produce undesirable audio effects when content shifts from a high overall loudness level to a lower overall loudness level. Such loudness drops may be detected, and the loudness control system may be adjusted to minimize the undesirable effects during the transition between loudness levels.

Abstract: There are disclosed automatic mixers and methods for creating a surround audio mix. A set of rules may be stored in a rule base. A rule engine may select a subset of the set of rules based, at least in part, on metadata associated with a plurality of stems. A mixing matrix may mix the plurality of stems in accordance with the selected subset of rules to provide three or more output channels.

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: Various exemplary embodiments relate to a method and apparatus for processing object-based audio signals to influence the reproduction of the audio signals. The apparatus may include an object-based audio decoder and an object post-processor. The apparatus is configured to receive an input audio stream comprising encoded dialog, music, effect, and height objects, decode from the input audio stream the dialog, music, and effect objects, process the dialog, music, and effect (DME) objects in separated signal paths; and mix the processed DME objects to produce an output audio signal for individual and customized rendering of the dynamic objects.

Abstract: A computing device may receive or otherwise access a base audio layer and one or more enhancement audio layers. The computing device can reconstruct the retrieved base layer and/or enhancement layers into a single data stream or audio file. The local computing device may process audio frames in a highest enhancement layer retrieved in which the data can be validated (or a lower layer if the data in audio frames in the enhancement layer(s) cannot be validated) and build a stream or audio file based on the audio frames in that layer.

Abstract: A multi-rate audio processing system and method provides real-time measurement and processing of amplitude/phase changes in the transition band of the lowest frequency subband caused by the audio processing that can be used to apply amplitude/phase compensation to the higher subband(s). Tone signals may be injected into the transition band to provide strong tonal content for measurement and processing. The real-time measurement and compensation adapts to time-varying amplitude/phase changes regardless of the source of the change (e.g. non-linear time-varying linear or user control parameters) and provides universal applicability for any linear audio processing.

Abstract: A bass management system and method for mitigating bass management errors by using explicit information available in the object audio rendering process and deriving the correct subwoofer contribution for each audio object. Embodiments of the bass management system and method are used to maintain the correct balance of the bass reproduced by the subwoofer relative to the sound coming out of the other speakers. The system and method are useful for a variety of different speaker configurations, including speaker configurations having different speaker sub-zones. Power-normalized gain coefficients for each speaker are combined and the power of the combined gain coefficients is computed and used to obtain a power-preserving subwoofer contribution coefficient. This subwoofer contribution coefficient is applied to the bass portion of the audio signal and audio objects to determine the contribution of a particular subwoofer.

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: Embodiments of systems and methods are described for generating layered audio such that computing devices can request a variable amount of data based on criteria such as their available bandwidth, device capability, or user selection. A base layer and one or more enhancement layers that incrementally enhance the previous layers may be generated. A computing device may retrieve the base layer and/or one or more enhancement layers, adjusting, in real-time or near real-time, which layers are retrieved based on fluctuations in the available bandwidth among other possible criteria.

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: 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: A method and apparatus may be used to perform personalized audio virtualization. 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 an audio signal, scale the audio signal, and perform a convolution and reverberation on the scaled audio signal to produce a convolved audio signal. The apparatus may be configured to filter the convolved audio signal and process the filtered audio signal for output.

Abstract: A method and apparatus may be used to perform personalized audio virtualization. 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 an audio signal, scale the audio signal, and perform a convolution and reverberation on the scaled audio signal to produce a convolved audio signal. The apparatus may be configured to filter the convolved audio signal and process the filtered audio signal for output.

Abstract: A method for calibrating a surround sound system is disclosed. The method utilizes a microphone array integrated in a front center loudspeaker of the surround sound system or a soundbar facing a listener. Positions of each loudspeaker relative to the microphone array can be estimated by playing a test signal at each loudspeaker and measuring the test signal received at the microphone array. The listener's position can also be estimated by receiving the listener's voice or other sound cues made by the listener using the microphone array. Once the positions of the loudspeakers and the listener's position are estimated, spatial calibrations can be performed for each loudspeaker in the surround sound system so that listening experience is optimized.

Abstract: A multi-rate audio processing system and method provides real-time measurement and processing of amplitude/phase changes in the transition band of the lowest frequency subband caused by the audio processing that can be used to apply amplitude/phase compensation to the higher subband(s). Tone signals may be injected into the transition band to provide strong tonal content for measurement and processing. The real-time measurement and compensation adapts to time-varying amplitude/phase changes regardless of the source of the change (e.g. non-linear time-varying linear or user control parameters) and provides universal applicability for any linear audio processing.

Abstract: A sound field coding system and method that provides flexible capture, distribution, and reproduction of immersive audio recordings encoded in a generic digital audio format compatible with standard two-channel or multi-channel reproduction systems. This end-to-end system and method mitigates any impractical need for standard multi-channel microphone array configurations in consumer mobile devices such as smart phones or cameras. The system and method capture and spatially encode two-channel or multi-channel immersive audio signals that are compatible with legacy playback systems from flexible multi-channel microphone array configurations.

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.