Abstract: One example may include a method that includes initiating an audio recording to capture audio data, comparing the audio data received from a microphone of a mobile device to an audio data range, determining whether the audio data is above an optimal level based on a result of the comparison, and queuing the audio data in an audio data queue when the audio data is above the optimal level.

Abstract: A method of recorded message detection is provided. In this an audio restream is received and a set of landmark features is identified in a section of the audio stream. From these landmark features an audio finger print for the section of the audio stream is determined. This audio finger print is compared with at least one of the plurality of stored audio finger prints, each derived from a respective audio stream. It is determined that the received audio stream is a recorded message if a derived audio finger print is substantially equivalent to one of the plurality of stored audio finger prints representing a recorded message.

Abstract: An audio encoding and decoding method includes obtaining a channel combination scheme for a current frame, obtaining an encoding mode of the current frame based on a downmix mode of a previous frame and the channel combination scheme for the current frame, performing time-domain downmix processing on left and right channel signals of the current frame based on the encoding mode of the current frame to obtain primary and secondary channel signals of the current frame, and encoding the primary and secondary channel signals of the current frame.

Abstract: A noise suppressor includes a memory; and a processor coupled to the memory, the processor is configured to: acquire a noise signal in a vicinity of a head region of an occupant seated in a vehicle seat; generate a control sound having an opposite phase to the acquired noise signal; acquire a sound signal having a predetermined frequency that is different from the noise signal; and output, as an output sound signal, the acquired sound signal in combination with the generated control sound.

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 multi-channel decorrelator for providing a plurality of decorrelated signals on the basis of a plurality of decorrelator input signals is configured to premix a first set of N decorrelator input signals into a second set of K decorrelator input signals, wherein K<N. The multi-channel decorrelator is configured to provide a first set of K? decorrelator output signals on the basis of the second set of K decorrelator input signals. The multi-channel decorrelator is further configured to upmix the first set of K? decorrelator output signals into a second set of N? decorrelator output signals, wherein N?>K?. The multi-channel decorrelator can be used in a multi-channel audio decoder. A multi-channel audio encoder provides complexity control information for the multi-channel decorrelator.

Abstract: A method and device for processing information are provided. The method for processing information includes: determining a current output voltage of a power supply in a terminal device; determining a target parameter of an audio processing circuit in the terminal device according to the current output voltage; and configuring the audio processing circuit for processing an audio signal according to the target parameter. With the embodiments of the present disclosure, an impact of a change of the output voltage of the power supply on output volume of a loudspeaker can be reduced, thereby improving user experience.

Abstract: An electronic apparatus includes an internal speaker, a communication unit configured to perform a wireless communication with a wearable external speaker, and a control unit. The control unit establishes wireless connection with the external speaker through the communication unit and the control unit continuously performs a control operation for preventing an output of an audio signal from the internal speaker in a case where the wireless connection with the external speaker is disconnected by a first factor in a state where the control unit performs the control operation. The control unit cancels performance of the control operation in a case where the wireless connection with the external speaker is disconnected by a second factor different from the first factor in a state where the control unit performs the control operation for preventing the output of the audio signal from the internal speaker.

Abstract: Wireless, wearable audio devices and methods for assigning position and operational controls. Wireless audio devices or earbuds are described that are provided as a pair in which both audio devices are physically substantially identical. In use, the audio devices are assigned a right or left position associated with the respective ear of the user in which they are to be disposed. Control surfaces are similarly assigned to provide particular control functionalities on a particular side of the user. Master and slave designations are provided and may be swapped based on a battery status of the respective audio devices. Battery usage and the duration of available use of the pair of audio devices may thus be maximized.

Abstract: A system for determining prioritisation values for two or more sounds within an audio clip includes: a feature extraction unit operable to extract characteristic features from the two or more sounds, a feature combination unit operable to generate a combined mix comprising extracted features from the two or more sounds, an audio assessment unit operable to identify the contribution of one or more of the features to the combined mix, a feature classification unit operable to assign a saliency score to each of the features in the combined mix, and an audio prioritisation unit operable to determine relative priority values for the two or more sounds in dependence upon the assigned saliency scores for each of one or more features of the sounds.

Abstract: Methods and systems for detecting and combining audio transmissions containing data are provided. In one embodiment, a method is provided that includes receiving multiple audio signals from multiple receivers. An audio transmission may be detected in a first subset of the audio signals and the audio transmission may contain data. A second subset of the audio signals may be identified with the strongest magnitude of the audio transmission. Portions of the second subset of the audio signals may then be combined to generate a combined audio transmission signal and the combined audio transmission signal may be decoded for further processing.

Abstract: Method to perform dynamic beamforming to reduce SNR in signals captured by head-wearable apparatus starts with microphones generating acoustic signals. Microphones are coupled to first stem of the apparatus and to second stem of the apparatus. First and second beamformers generate first and second beamformer signals, respectively. Noise suppressor attenuates noise content from the first beamformer signal and the second beamformer signal. Noise content from first beamformer signal are acoustic signals not collocated in second beamformer signal and noise content from second beamformer signal are acoustic signals not collocated in first beamformer signal. Speech enhancer generates clean signal comprising speech content from first noise-suppressed signal and second noise-suppressed signal. Speech content are acoustic signals collocated in first beamformer signal and second beamformer signal.

Abstract: Systems, methods, and software are disclosed herein for compressing audio data. In an implementation, sampled values of an audio signal have a dynamic range. A division of the dynamic range, into at least a lower range and an upper range, is identified based on a fixed mapping of a lower portion of the sampled values to a subset of quanta in a set of quanta having a depth less than a depth of the sampled values. Then an adaptive mapping of an upper portion of the sampled values to a remaining subset of quanta in the set of quanta is also identified, based at least on a dimension of the upper range. The fixed mapping is used to encode the lower portion of the sampled values based, while the adaptive mapping is used to encode the upper portion of the sampled values based on the adaptive mapping.

Abstract: In general, techniques are described by which to code scaled spatial components. A device comprising a memory and one or more processors may be configured to perform the techniques. The memory may store a bitstream including an encoded foreground audio signal and a corresponding quantized spatial component. The one or more processors may perform psychoacoustic audio decoding with respect to the encoded foreground audio signal to obtain a foreground audio signal, and determine, when performing psychoacoustic audio decoding, a bit allocation for the encoded foreground audio signal. The one or more processors may dequantize the quantized spatial component to obtain a scaled spatial component, and descale, based on the bit allocation, the scaled spatial component to obtain a spatial component. The one or more processors may reconstruct, based on the foreground audio signal and the spatial component, scene-based audio data.

Abstract: Provided is an information processing device including a control section that recognizes an intention of a first action subject on a basis of an action performed by the first action subject in a first context, generates information indicating the intention of the first action subject in a second context of the second action subject, and causes the information to be outputted to the second action subject. The first action subject is one of a user or a communication target of the user. The second action subject is another of the user or the communication target of the user.

Abstract: An audio encoding and decoding method and a related apparatus are provided. The audio encoding method includes: determining a channel combination scheme for a current frame; when the channel combination scheme for the current frame is different from a channel combination scheme for a previous frame, performing segmented time-domain downmix processing on left and right channel signals in the current frame based on the channel combination scheme for the current frame and the channel combination scheme for the previous frame, to obtain a primary channel signal and a secondary channel signal in the current frame; and encoding the obtained primary channel signal and secondary channel signal in the current frame.

Abstract: A voice output corresponding to a fixed position of a wide viewing angle image is easily obtained. A transmission unit configured to transmit spatial voice data and information regarding a predetermined number of registered viewpoints is included. For example, the spatial voice data is data of scene-based audio. Then, for example, the data of the scene-based audio is each component of an HoA format. For example, the information regarding a viewpoint includes information regarding an azimuth angle (azimuth information) and an elevation angle (elevation angle information) that indicate a position of this viewpoint. For example, the data of the scene-based audio and the information regarding the predetermined number of registered viewpoints are transmitted with being included in a packet of object audio.

Abstract: An apparatus, system and method for a portable microphone array system comprising a computing device and a case having an array of microphones embedded or integrated into the case. A user may position the laptop and case facing the general direction of a target audio source to capture a target acoustic audio input at the microphone array. The microphone array may deliver a first stage of beamformed audio from the acoustic audio input to the computing device via a communications interface or bus. The computing device may comprise an audio processor configured to perform one or more successive audio processing steps to process the audio input and render a digital audio output. The digital audio output may be outputted from the computing device to an audio output device, such as headphones or an earpiece.

Abstract: The present document describes a method (400) for encoding a soundfield representation (SR) input signal (101, 301) describing a soundfield at a reference position, wherein the SR input signal (101, 301) comprises a plurality of channels for a plurality of different directivity patterns of the soundfield at the reference position. The method (400) comprises extracting (401) one or more audio objects (103, 303) from the SR input signal (101, 301). Furthermore, the method (400) comprises determining (402) a residual signal (102, 302) based on the SR input signal (101, 301) and based on the one or more audio objects (103, 303). The method (400) also comprises performing joint coding of the one or more audio objects (103, 303) and/or the residual signal (102, 302). In addition, the method (400) comprises generating (403) a bitstream (701) based on data generated in the context of joint coding of the one or more audio objects (103, 303) and/or the residual signal (102, 302).

Abstract: A metamaterial comprising, a plurality of acoustic vector field sensors, each configured to sense an acoustic vector field of a fluid within a fluid-filled space in response to fluid waves, and producing an electrical signal corresponding to the sensed acoustic vector field; a processor configured to perform a time and space transform on the electrical signal; and at least one phased array transducer, configured to emit fluid waves according to a produced acoustic vector field pattern dependent on a result of the time and space transform, a within a portion of the fluid.