According to an aspect of the present invention, a method for reconstructing an audio signal having a baseband portion and a highband portion is disclosed. The method includes obtaining a decoded baseband audio signal by decoding an encoded audio signal and obtaining a plurality of subband signals by filtering the decoded baseband audio signal. The method further includes generating a high-frequency reconstructed signal by copying a number of consecutive subband signals of the plurality of subband signals and obtaining an envelope adjusted high-frequency signal. The method further includes generating a noise component based on a noise parameter. Finally, the method includes adjusting a phase of the high-frequency reconstructed signal and obtaining a time-domain reconstructed audio signal by combining the decoded baseband audio signal and the combined high-frequency signal to obtain a time-domain reconstructed audio signal.

An audio metadata providing apparatus and method and a multichannel audio data playback apparatus and method to support a dynamic format conversion are provided. Dynamic format conversion information may include information about a plurality of format conversion schemes that are used to convert a first format set by an author of multichannel audio data into a second format that is based on a playback environment of the multichannel audio data and that are each set for corresponding playback periods of the multichannel audio data. The audio metadata providing apparatus may provide audio metadata including the dynamic format conversion information. The multichannel audio data playback apparatus may identify the dynamic format conversion information from the audio metadata, may convert the first format of the multichannel audio data into the second format based on the identified dynamic format conversion information, and may play back the multichannel audio data in the second format.

Apparatus for processing an audio scene representing a sound field, the audio scene having information on a transport signal and a first set of parameters. The apparatus has a parameter processor for processing the first set of parameters to obtain a second set of parameters, wherein the parameter processor is configured to calculate at least one raw parameter for each output time frame using at least one parameter of the first set of parameters for the input time frame, to calculate a smoothing information such as a factor for each raw parameter in accordance with a smoothing rule, and to apply a corresponding smoothing information to the corresponding raw parameter to derive the parameter of the second set of parameters for the output time frame. The apparatus further has an output interface for generating a processed audio scene using the second set of parameters and the information on the transport signal.

Embodiments are disclosed for channel-based audio (CBA) (e.g., 22.2-ch audio) to object-based audio (OBA) conversion. The conversion includes converting CBA metadata to object audio metadata (OAMD) and reordering the CBA channels based on channel shuffle information derived in accordance with channel ordering constraints of the OAMD. The OBA with reordered channels is rendered in a playback device using the OAMD or in a source device, such as a set-top box or audio/video recorder. In an embodiment, the CBA metadata includes signaling that indicates a specific OAMD representation to be used in the conversion of the metadata. In an embodiment, pre-computed OAMD is transmitted in a native audio bitstream (e.g., AAC) for transmission (e.g., over HDMI) or for rendering in a source device. In an embodiment, pre-computed OAMD is transmitted in a transport layer bitstream (e.g., ISO BMFF, MPEG4 audio bitstream) to a playback device or source device.

Methods and systems for optimizing a routing of audio data to audio transmitting devices using a Bluetooth network are disclosed. One method includes receiving an encoded audio bitstream at a first speaker of the audio rendering system comprising a first and a second audio channels, separating a first set of spectral components of the first audio channel and a second set of spectral components of the second audio channel from the encoded audio bitstream, without decoding the audio bitstream, generating a first encoded bitstream from the first set of spectral components, and forwarding the first encoded bitstream to a second speaker of the audio rendering system over the wireless link.

An apparatus including circuitry configured to: obtain a binaural audio signal; obtain, based on the binaural audio signal, at least one direction parameter of at least one frequency band of the binaural audio signal; process the binaural audio signal to generate at least two audio signals for loudspeaker reproduction by modifying an inter-channel difference of the at least one frequency band of the binaural audio signal based on the at least one direction parameter for the at least one frequency band; and output the at least two audio signals for loudspeaker reproduction.

A device includes one or more processors configured to generate, at an encoder portion of an autoencoder, first output data at least partially based on first input data and to generate, at a decoder portion or the autoencoder, a representation of the first input data at least partially based on the first output data. The one or more processors are configured to generate, at the encoder portion, second output data based on second input data and first state data and to generate, at the decoder portion, a representation of the second input data based on the second output data and second state data. Each of the first state data and the second state data correspond to the state of the decoder portion resulting from generation of the representation of the first input data. The first and second input data correspond to sequential values of a signal to be encoded.

The disclosure provides a speech processing method and a device thereof. The method includes: acquiring a speech sampling signal frame in a mixed-excitation linear prediction (MELP) speech coding system and estimating signal quality of the speech sampling signal frame; determining, based on the signal quality, a specific linear prediction coding (LPC) order used by an LPC circuit; controlling the LPC circuit to convert the speech sampling signal frame into a line spectrum pair parameter based on the specific LPC order; replacing a speech signal spectrum of the speech sampling signal frame with the line spectrum pair parameter to generate a predicted speech signal; and performing a speech coding operation and a signal synthesizing operation of the MELP speech coding system based on the predicted speech signal.

A first playback device is configured to: operate as part of a synchrony group that comprises the first playback device and a second playback device; obtain a first version of audio content that is encoded according to a first encoding format; determine that the first version of the audio content is unsuitable for playback by the second playback device; based on the determination, (i) decode the first version of the audio content and (ii) re-encode a second version of the audio content according to a second encoding format; transmit the second version of the audio content to the second playback device for playback; cause the second playback device to play back the second version of the audio content; and play back the first version of the audio content in synchrony with the playback of the second version of the audio content by the second playback device.

A device-management system performs processing, such as audio processing, in an instance of a virtual machine corresponding to a functionally limited (local) device. To register the local device, the device-management system receives a registration request that includes device information, encryption data, and an indication of an associated user account. The device-management system then sends this registration data to a service-provider system, which returns a shared encryption key. The device-management system and the local device may use this shared encryption key to securely communicate. The device-management system may de-allocate the instance upon detecting a period of inactivity of the local device and may re-allocate the instance when new activity is detected. The device-management system may further determine when and if audio data to be sent to the local device is encoded using a codec not implemented by the local device. Upon this determination, the device-management system may transcode the audio data such that is encoded using a known codec.