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: An apparatus including circuitry configured for determining for at least one first audio signal of an audio signal format, at least one metadata parameter; determining for at least one further audio signal of a further audio signal format; at least one further metadata parameter; controlling combining of the at least one metadata parameter with the at least one further metadata parameter to generate a combined metadata, wherein the combined metadata is configured to be associated with a combined audio signal formed from the at least one first audio signal and the at least one further audio signal in such a way that the combined metadata includes at least one spatial audio parameter.

Abstract: In an example embodiment, a natural language processing (NLP) machine learning and rules-based text extraction and analysis approach is used to convert a textual service recommendation document into customer-tailored actions considering the specific context-based executable script. This creates end-to-end automation in implementing suggestions provided in textual documents. Actions mentioned in the document can be processed automatically whenever possible, or at least transformed into a semi-automated action with system support. The solution can be configured to automate end-to-end converting of documents into personalized technical scripts and implementing these scripts at the customer-side.

Abstract: Disclosed are a system, method and apparatus to generate service codes based, at least in part, on electronic documents. In an embodiment, tokens may be embedded in an electronic document based, at least in part, on a linguistic analysis of the electronic document. Likelihoods of applicability of service codes to the electronic document may be determined based, at least in part, on the embedding of tokens.

Abstract: Audio encoder for encoding audio input data to obtain audio output data includes an input interface for receiving a plurality of audio channels, a plurality of audio objects and metadata related to one or more of the plurality of audio objects; a mixer for mixing the plurality of objects and the plurality of channels to obtain a plurality of pre-mixed channels, each pre-mixed channel including audio data of a channel and audio data of at least one object; a core encoder for core encoding core encoder input data; and a metadata compressor for compressing the metadata related to the one or more of the plurality of audio objects, wherein the audio encoder is configured to operate in at least one mode of the group of two modes.

Abstract: In multichannel audio coding, improved computational efficiency is achieved by computing comparison parameters for ITD compensation between any two channels in the frequency domain for a parametric audio encoder. This may mitigate negative effects on encoder parameter estimates.

Abstract: A system, method and computer-readable storage device provides an improved speech processing approach in which hyper parameters used for speech recognition are modified dynamically or in batch mode rather than fixed statically. The method includes estimating, via a model trained on audio data and/or metadata, a set of parameters useful for performing automatic speech recognition, receiving speech at an automatic speech recognition system, applying, by the automatic speech recognition system, the set of parameters to processing the speech to yield text and outputting the text from the automatic speech recognition system.

Abstract: Techniques are described herein for detecting and suppressing commands in media that may trigger another automated assistant. A method includes: determining, for each of a plurality of automated assistant devices in an environment that are each executing at least one automated assistant, an active capability of the automated assistant device; initiating playback of digital media by an automated assistant; in response to initiating playback, processing the digital media to identify an audio segment in the digital media that, upon playback, is expected to trigger activation of at least one automated assistant executing on at least one of the plurality of automated assistant devices in the environment, based on the active capability of the at least one of the plurality of automated assistant devices; and in response to identifying the audio segment in the digital media, modifying the digital media to suppress the activation of the at least one automated assistant.

Abstract: An electronic apparatus is provided. The electronic apparatus includes a microphone, a communication interface including circuitry and a processor configured to, based on identifying that a trigger word is included in a first sound signal received through the microphone, enter a voice recognition mode, identify a gain value for adjusting an intensity of the first sound signal to be in a predetermined intensity range based on the intensity of the first sound, adjust an intensity of a second sound signal received through the microphone in the voice recognition mode based on the identified gain value, and control the communication interface to transmit a user command obtained based on voice recognition regarding the adjusted second sound signal, to an external apparatus.

Abstract: An example audio decoding device includes a memory configured to store at least a portion of a coded audio bitstream; and one or more processors configured to: decode, based on the coded audio bitstream, a representation of a soundfield; decode, based on the coded audio bitstream, a syntax element indicating a selection of either a head-related transfer function (HRTF) or a binaural room impulse response (BRIR); and render, using the selected HRTF or BRIR, speaker feeds from the soundfield.

Abstract: A training system produces a resource-efficient machine-trained model via a training architecture that employs plural processing paths. Some of the processing paths incorporate the use of auxiliary information that imparts external knowledge about source items being processed. The training architecture also employs contrastive learning that operates at different respective levels within the training architecture. For instance, the training architecture uses encoder-level contrastive learning to compare output information generated by different encoders within the training architecture. The training architecture uses decoder-level contrastive learning to compare output information produced by different decoders within the training architecture. An inference-stage system performs an application task using the model produced by the training system.

Abstract: Systems and methods for enhancing a headset user's own voice include at least two outside microphones, an inside microphone, audio input components operable to receive and process the microphone signals, a voice activity detector operable to detect speech presence and absence in the received and/or processed signals, and a cross-over module configured to generate an enhanced voice signal. The audio processing components includes a low frequency branch comprising low pass filter banks, a low frequency spatial filter, a low frequency spectral filter and an equalizer, and a high frequency branch comprising highpass filter banks, a high frequency spatial filter, and a high frequency spectral filter.

Abstract: An electronic apparatus is provided. The electronic apparatus includes a microphone, a communication interface including circuitry and a processor configured to, based on identifying that a trigger word is included in a first sound signal received through the microphone, enter a voice recognition mode, identify a gain value for adjusting an intensity of the first sound signal to be in a predetermined intensity range based on the intensity of the first sound, adjust an intensity of a second sound signal received through the microphone in the voice recognition mode based on the identified gain value, and control the communication interface to transmit a user command obtained based on voice recognition regarding the adjusted second sound signal, to an external apparatus.

Abstract: An active noise control system selects one reference sensor providing a reference signal with the largest coherence to a noise signal, among a plurality of available reference sensors, as a first entry of a reference sensor set. After selecting the first entry, the active noise control system repeats a process in which a sensor capable of providing the largest information quantity to a current reference sensor set among remaining sensors is selected as a new entry of a reference sensor set, until a desired number of sensors is reached or a desired control level is reached. When the reference sensor set is determined, the active noise control system utilizes the entries of the reference sensor set to generates a noise control signal suitable for canceling the noise signal.

Abstract: A method, computer program product, and computing system for generating a plurality of acoustic relative transfer functions for a plurality of audio acquisition devices of an audio recording system deployed in an acoustic environment. The plurality of acoustic relative transfer functions may be encoded into a first embedding of acoustic relative transfer functions and at least a second embedding of acoustic relative transfer functions. Information may be extracted from at least the first embedding of acoustic relative transfer functions.

Abstract: An audio system includes an audio/video receiver, a power supply/wireless audio distribution assembly connected to the audio/video receiver, speaker wire, and speakers compatible with the power supply/wireless audio distribution assembly.

Abstract: Multichannel audio playback devices and associated systems and methods are disclosed herein. In some examples, a first playback device is configured to receive a source stream of audio content comprising left, right and center input channels. In a first mode, the first playback device is configured to play back audio via a plurality of transducers based on the left, right, and center input channels. In a second mode, in which the first playback device is bonded to second and third playback devices, the first playback device is configured to (i) play back audio via the plurality of transducers based on at least the center input channel, (ii) cause audio to be played via the second playback device based on at least the right input channel, and (iii) cause audio to be played via the third playback device based on at least the left input channel.

Abstract: A method is described which decodes a downmix matrix for mapping a plurality of input channels of audio content to a plurality of output channels, the input and output channels being associated with respective speakers at predetermined positions relative to a listener position, wherein the downmix matrix is encoded by exploiting the symmetry of speaker pairs of the plurality of input channels and the symmetry of speaker pairs of the plurality of output channels. Encoded information representing the encoded downmix matrix is received and decoded for obtaining the decoded downmix matrix.

Abstract: A method, computer program product, and computing system for selecting a reference audio acquisition device from a plurality of audio acquisition devices of an audio recording system. Audio encounter information of the reference microphone may be encoded, thus defining encoded reference audio encounter information. A plurality of acoustic relative transfer functions between the reference microphone and the plurality of audio acquisition devices of the audio recording system may be generated. The encoded reference audio encounter information and a representation of the plurality of acoustic relative transfer functions may be transmitted.

Abstract: A multi-channel transformer acoustic model that processes a plurality of audio signals output by microphones of a microphone array and outputs probabilities for acoustic units of an utterance represented in the audio signals. The audio signals represent the individual microphones' respective capturing of the utterance. The multi-channel model may perform self-attention on embeddings of the audio signals and then cross-channel attention across the attended audio signals. The cross-channel attention may involve processing of signals relative to each other to model the relationships across channels within and across time frames. The multi-channel model may include a transducer to perform processing frame-by-frame.