Abstract: Embodiments of systems and methods are described for reducing undesired leakage energy produced by a non-front-facing speaker in a multi-speaker system. For example, the multi-speaker system can include an array of forward-facing speakers, one or more upward-facing speakers, and/or one or more side-facing speakers. Filters coupled to any two of the speakers in the multi-speaker system can generate audio signals output by the coupled speakers to reduce, attenuate, or cancel a portion of an audio signal output by one or more non-front-facing speakers that acoustically propagates along a direct path from the respective non-front-facing speaker to a listening position in a listening area in front of the multi-speaker system.

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 digital signal is processed by splitting it into at least two frequency subbands and the two subband signals are downsampled. A filter is applied in at least one of the subband signals. At least one of the phase and magnitude of the subband filtered signals is matched in the transition frequency band between the two subbands.

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 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: 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: Methods and systems of reproducing object-based audio are disclosed. In some embodiments, vector base amplitude panning (VBAP) is used for playing back an object's audio. Using the positioning of sound reproduction devices and object's location information, rendering can determine which sound reproduction devices are used for playing back the object's audio. For example, a triangle in which the object is positioned at a given time can be identified. The triangle can have sound reproduction devices as vertices, and the object's audio can be rendered on the sound reproduction devices corresponding to the vertices of the triangle. In some embodiments, ambiguities associated with VBAP-based rendering are identified and resolved.

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 method and apparatus for processing object-based audio signals is provided. The apparatus receives a plurality of object-based audio signals. Each object-based audio signal of the object-based audio signals includes audio waveform data and object metadata associated with the audio waveform data. The object metadata includes at least one of a loudness parameter or a power parameter associated with the audio waveform data. The apparatus determines a loudness metric based on the received object-based audio signals and based on the at least one of the loudness parameter or the power parameter for each object-based audio signal of the received object-based audio signals. In one configuration, the apparatus renders the received object-based audio signals to a set of output signals based on the determined loudness metric. In another configuration, the apparatus transmits (e.g., broadcast, file delivery, or streaming) the received object-based audio signals based on the determined loudness metric.

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: An audio signal is processed in the frequency domain to convert an input signal format to an output signal format. That is, a multichannel audio signal intended for playback over a predefined speaker layout can be formatted to achieve spatial reproduction over a different layout comprising a different number of speakers.

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.

Abstract: An audio processing system for processing a single channel audio signal includes a a processor configured to derive a synthetic difference component from the single channel audio input signal a filtering module configured to apply a first filter to the sum signal represented by the single channel signal and to apply a second filter to the synthetic difference signal; and a control module configured to crossfade to control the amount of the resulting audio signal effect by respectively scaling the sum signal and the difference signal.

Abstract: A two-channel phase-amplitude stereo encoding and decoding scheme enabling flexible and spatially accurate interactive 3-D audio reproduction via standard audio-only two-channel transmission. The encoding scheme allows associating a 2-D or 3-D positional localization to each of a plurality of sound sources by use of frequency independent inter-channel phase and amplitude differences. The decoder is based on frequency-domain spatial analysis of 2-D or 3-D directional cues in a two-channel stereo signal and re-synthesis of these cues using any preferred spatialization technique, thereby allowing faithful reproduction of positional audio cues and reverberation or ambient cues over arbitrary multi-channel loudspeaker reproduction formats or over headphones, while preserving source separation despite the intermediate encoding over only two audio channels.

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: The present invention describes techniques that can be used to provide novel methods of spatial audio rendering using adapted M-S matrix shuffler topologies. Such techniques include headphone and loudspeaker-based binaural signal simulation and rendering, stereo expansion, multichannel upmix and pseudo multichannel surround rendering.

Abstract: Methods and systems of reproducing object-based audio are disclosed. In some embodiments, vector base amplitude panning (VBAP) is used for playing back an object's audio. Using the positioning of sound reproduction devices and object's location information, rendering can determine which sound reproduction devices are used for playing back the object's audio. For example, a triangle in which the object is positioned at a given time can be identified. The triangle can have sound reproduction devices as vertices, and the object's audio can be rendered on the sound reproduction devices corresponding to the vertices of the triangle. In some embodiments, ambiguities associated with VBAP-based rendering are identified and resolved.