Abstract: Enclosed are embodiments for audio processing that combines complementary aspects of Spatial Reconstruction (SPAR) and Directional Audio Coding (DirAC) technologies, including higher audio quality, reduced bitrate, input/output format flexibility and/or reduced computational complexity, to produce a codec (e.g., an Ambisonics codec) that has better overall performance than DirAC or SPAR codecs.

Abstract: The present document describes a method for generating a bitstream, wherein the bitstream comprises a sequence of superframes for a sequence of frames of an immersive audio signal. The method comprises, repeatedly for the sequence of superframes, inserting coded audio data for one or more frames of one or more downmix channel signals derived from the immersive audio signal, into data fields of a superframe; and inserting metadata for reconstructing one or more frames of the immersive audio signal from the coded audio data, into a metadata field of the superframe.

Abstract: The disclosed embodiments enable converting audio signals captured in various formats by various capture devices into a limited number of formats that can be processed by an audio codec (e.g., an Immersive Voice and Audio Services (IVAS) codec). In an embodiment, a simplification unit of the audio device receives an audio signal captured by one or more audio capture devices coupled to the audio device. The simplification unit determines whether the audio signal is in a format that is supported/not supported by an encoding unit of the audio device. Based on the determining, the simplification unit, converts the audio signal into a format that is supported by the encoding unit. In an embodiment, if the simplification unit determines that the audio signal is in a spatial format, the simplification unit can convert the audio signal into a spatial “mezzanine” format supported by the encoding.

Abstract: An aspect of the present disclosure relates to processing audio comprising decoding a first bitstream (b1) to obtain decoded immersive audio content (A), decoding a second bitstream (bp) to obtain pose information (P, V, V?) associated with a user of a lightweight processing device, determining a first head-pose (P?) based on the pose information, providing a downmix representation (Dmx) of the immersive audio content (A) corresponding to the first head pose (P?), rendering a set of binaural representations (BINn) of the immersive audio content (A), wherein the binaural representations correspond to a second set of head poses (Pn), computing reconstruction metadata (M) to enable reconstruction of the set of binaural representations from the downmix representation (Dmx), the metadata (M) including the first head pose (P?), and encoding the downmix representation (Dmx) and the reconstruction metadata (M) in a third bitstream (b2).

Abstract: Embodiments are disclosed for bitrate distribution in immersive voice and audio services. In an embodiment, a method of encoding an IVAS bitstream comprises: receiving an input audio signal; downmixing the input audio signal into one or more downmix channels and spatial metadata; reading a set of one or more bitrates for the downmix channels and a set of quantization levels for the spatial metadata from a bitrate distribution control table; determining a combination of the one or more bitrates for the downmix channels; determining a metadata quantization level from the set of metadata quantization levels using a bitrate distribution process; quantizing and coding the spatial metadata using the metadata quantization level; generating, using the combination of one or more bitrates, a downmix bitstream for the one or more downmix channels; combining the downmix bitstream, the quantized and coded spatial metadata and the set of quantization levels into the IVAS bitstream.

Abstract: The disclosed embodiments enable converting audio signals captured in various formats by various capture devices into a limited number of formats that can be processed by an audio codec (e.g., an Immersive Voice and Audio Services (IVAS) codec). In an embodiment, a simplification unit of the audio device receives an audio signal captured by one or more audio capture devices coupled to the audio device. The simplification unit determines whether the audio signal is in a format that is supported/not supported by an encoding unit of the audio device. Based on the determining, the simplification unit, converts the audio signal into a format that is supported by the encoding unit. In an embodiment, if the simplification unit determines that the audio signal is in a spatial format, the simplification unit can convert the audio signal into a spatial “mezzanine” format supported by the encoding.

Abstract: Methods and systems for improving signal processing by smoothing the covariance matrix of a multi-channel signal by setting a forgetting factor based on the bins of a band. A method and system for resetting the smoothing based on transient detection is also disclosed. A method and system for resampling for the smoothing during a banding transition is also disclosed.

Abstract: The present document describes a method (500) for generating a bitstream (101), wherein the bitstream (101) comprises a sequence of superframes (400) for a sequence of frames of an immersive audio signal (111). The method (500) comprises, repeatedly for the sequence of superframes (400), inserting (501) coded audio data (206) for one or more frames of one or more downmix channel signals (203) derived from the immersive audio signal (111), into data fields (411, 421, 412, 422) of a superframe (400); and inserting (502) metadata (202, 205) for reconstructing one or more frames of the immersive audio signal (111) from the coded audio data (206), into a metadata field (403) of the superframe (400).

Abstract: The disclosed embodiments enable converting audio signals captured in various formats by various capture devices into a limited number of formats that can be processed by an audio codec (e.g., an Immersive Voice and Audio Services (IVAS) codec). In an embodiment, a simplification unit of the audio device receives an audio signal captured by one or more audio capture devices coupled to the audio device. The simplification unit determines whether the audio signal is in a format that is supported/not supported by an encoding unit of the audio device. Based on the determining, the simplification unit, converts the audio signal into a format that is supported by the encoding unit. In an embodiment, if the simplification unit determines that the audio signal is in a spatial format, the simplification unit can convert the audio signal into a spatial “mezzanine” format supported by the encoding.

Abstract: An apparatus and method of excursion protection of a loudspeaker. The method includes attenuating selected bands in a transform domain, controlled by a feedback signal resulting from an excursion transfer function that has been modified according to the real-time operational characteristics of the loudspeaker. In this manner, the system reduces the amount of wideband attenuation needed to address the predicted excursion, resulting in a better listening experience.

Abstract: Systems, methods, and computer program products of audio processing based on Adaptive Intermediate Spatial Format (AISF) are described. The AISF is an extension to ISF that allows spatial resolution around an ISF ring to be adjusted dynamically with respect to content of incoming audio objects. An AISF encoder device adaptively warps each ISF ring during ISF encoding to adjust angular distance between objects, resulting in increase in uniformity of energy distribution around the ISF ring. At an AISF decoder device, matrices that decode sound positions to the output speaker take into account the warping that was performed at the AISF encoder device to reproduce the true positions of sound sources.

Abstract: The present document relates to methods and apparatus for rendering input audio for playback in a playback environment. The input audio includes at least one audio object and associated metadata, and the associated metadata indicates at least a location of the audio object. A method for rendering input audio including divergence metadata for playback in a playback environment comprises creating two additional audio objects associated with the audio object such that respective locations of the two additional audio objects are evenly spaced from the location of the audio object, on opposite sides of the location of the audio object when seen from an intended listener's position in the playback environment, determining respective weight factors for application to the audio en.

Abstract: Systems and methods for manual characterization of perceived transducer distortion are described. The system includes a signal generator, a controller, a multi-band compressor, and an audio transducer. The signal generator is configured to generate a test signal for a frequency band. The test signal includes at least two simultaneous frequency-modulated tones, which may be combined together in an amplitude-modulated envelope. The at least two simultaneous frequency-modulated tones have different frequencies within the frequency band. The controller is configured to receive user input indicating a gain value. The multi-band compressor is coupled to the signal generator. The multi-band compressor is configured to adjust an amplitude of a component of the test signal based on the gain value. The audio transducer is coupled to the multi-band compressor. The audio transducer is configured to generate an audio signal based on the test signal.

Abstract: Systems and methods for automatic characterization of perceived transducer distortion are described. The system includes a controller configured to receive a distortion level; a signal generator configured to generate a test signal for a frequency band in response to the distortion level, the test signal including at least two simultaneous tones, the at least two simultaneous tones having different frequencies within the frequency band; an audio transducer configured to generate an audio signal based on the test signal; and a distortion tuner configured to receive the audio signal and to determine the distortion level of the system based on a detected amount of distortion in the audio signal.

Abstract: A method, apparatus, and medium for rendering an audio program to a number of loudspeaker feed signals are provided. The audio program may include one or more audio objects, and metadata associated with each of the one or more audio objects. The metadata may include position information indicating a time-varying position of the audio object and a parameter indicating whether the audio object should be reproduced at the time-varying position, or at one of a plurality of fixed positions. In response to the position and the parameter, a position at which to reproduce each audio object may be determined. The determined position may be one of the plurality of fixed positions that is nearest to the time-varying position indicated by the position information. Each audio object may be reproduced at the determined position by rendering the audio object into one or more of the loudspeaker feed signals.

Abstract: Systems, methods, and computer program products of audio processing based on Adaptive Intermediate Spatial Format (AISF) are described. The AISF is an extension to ISF that allows spatial resolution around an ISF ring to be adjusted dynamically with respect to content of incoming audio objects. An AISF encoder device adaptively warps each ISF ring during ISF encoding to adjust angular distance between objects, resulting in increase in uniformity of energy distribution around the ISF ring. At an AISF decoder device, matrices that decode sound positions to the output speaker take into account the warping that was performed at the AISF encoder device to reproduce the true positions of sound sources.

Abstract: A device (160) and a method are proposed for tuning a frequency-dependent attenuation stage (122),with the purpose of suppressing non-linear distortion occurring in an audio reproduction system (120) associated with the frequency-dependent attenuation stage. The device comprises a receiving section (162) adapted to receive data representing an output acoustic signal (140) from the audio reproduction system,recorded upon excitation of the audio reproduction system by a predetermined input data signal (110)a first distortion detection section (163) adapted to detect presence of non-linear distortion based on the received data and to apply psycho-acoustic compensation to the detected non-linear distortion and a control section (164) adapted to determine, based on the psycho-acoustically compensated non-linear distortion, control information (170) suitable for controlling the frequency-dependent attenuation stage.