An apparatus for generating a description of a combined audio scene, includes: an input interface for receiving a first description of a first scene in a first format and a second description of a second scene in a second format, wherein the second format is different from the first format; a format converter for converting the first description into a common format and for converting the second description into the common format, when the second format is different from the common format; and a format combiner for combining the first description in the common format and the second description in the common format to obtain the combined audio scene.

An apparatus for generating a description of a combined audio scene, includes: an input interface for receiving a first description of a first scene in a first format and a second description of a second scene in a second format, wherein the second format is different from the first format; a format converter for converting the first description into a common format and for converting the second description into the common format, when the second format is different from the common format; and a format combiner for combining the first description in the common format and the second description in the common format to obtain the combined audio scene.

Methods, an encoder and a decoder are configured for transition between frames with different internal sampling rates. Linear predictive (LP) filter parameters are converted from a sampling rate S1 to a sampling rate S2. A power spectrum of a LP synthesis filter is computed, at the sampling rate S1, using the LP filter parameters. The power spectrum of the LP synthesis filter is modified to convert it from the sampling rate S1 to the sampling rate S2. The modified power spectrum of the LP synthesis filter is inverse transformed to determine autocorrelations of the LP synthesis filter at the sampling rate S2. The autocorrelations are used to compute the LP filter parameters at the sampling rate S2.

A technique is provided that can reduce degradation of the sound quality due to a tandem connection of paired coding and decoding, and can reduce the operation processing amount and the required memory amount of a multipoint control unit. In multipoint connection between terminals of a plurality of communication networks (for example, a fixed phone line and a mobile phone line) having different communication capacities, when a multichannel coding including a monaural coding scheme of a communication network having a smaller communication capacity is used in a communication network having a larger communication capacity to transmit sounds of a plurality points to the communication network terminal having the larger communication capacity, control is exercised such that monaural codes of the plurality points are output.

Disclosed is an audio signal processing apparatus for rendering an input audio signal. The audio signal processing apparatus may include a processor configured to obtain an input audio signal including an ambisonics signal and a non-diegetic channel difference signal, render the ambisonics signal to generate a first output audio signal, mix the first output audio signal and the non-diegetic channel difference signal to generate a second output audio signal, and output the second output audio signal.

A technique is provided that can reduce degradation of the sound quality due to a tandem connection of paired coding and decoding, and can reduce the operation processing amount and the required memory amount of a multipoint control unit. In multipoint connection between terminals of a plurality of communication networks (for example, a fixed phone line and a mobile phone line) having different communication capacities, when a multichannel coding including a monaural coding scheme of a communication network having a smaller communication capacity is used in a communication network having a larger communication capacity to transmit sounds of a plurality points to the communication network terminal having the larger communication capacity, control is exercised such that monaural codes of the plurality points are output.

A technique is provided that can reduce degradation of the sound quality due to a tandem connection of paired coding and decoding, and can reduce the operation processing amount and the required memory amount of a multipoint control unit. In multipoint connection between terminals of a plurality of communication networks (for example, a fixed phone line and a mobile phone line) having different communication capacities, when a multichannel coding including a monaural coding scheme of a communication network having a smaller communication capacity is used in a communication network having a larger communication capacity to transmit sounds of a plurality points to the communication network terminal having the larger communication capacity, control is exercised such that monaural codes of the plurality points are output.

Described herein is an audio decoder for decoding a bitstream of encoded audio data, wherein the bitstream of encoded audio data represents a sequence of audio sample values and comprises a plurality of frames, wherein each frame comprises associated encoded audio sample values, the audio decoder comprising: a determiner configured to determine whether a frame of the bitstream of encoded audio data is an immediate playout frame comprising encoded audio sample values associated with a current frame and additional information; and an initializer configured to initialize the decoder if the determiner determines that the frame is an immediate playout frame, wherein initializing the decoder comprises decoding the encoded audio sample values comprised by the additional information before decoding the encoded audio sample values associated with the current frame. Described are further a method for decoding said bitstream of encoded audio data as well as an audio encoder, a system of audio encoders and a method for generating said bitstream of encoded audio data with immediate playout frames. Described are moreover also an apparatus for generating immediate playout frames in a bitstream of encoded audio data or for removing immediate playout frames from a bitstream of encoded audio data and respective non-transitory digital storage media.

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.

Disclosed are an audio rate conversion system and an electronic apparatus. The audio rate conversion system includes an integrator-comb filter, a multi-rate filter and a first half-band filter, an input of the integrator-comb filter being accessed with digital audio data, an output of the integrator-comb filter being sequentially connected to the multi-rate filter and the first half-band filter; where, the integrator-comb filter is configured to reduce a rate of the digital audio data according to a preset decimation rate; the multi-rate filter is configured to convert a rate of digital audio data output by the integrator-comb filter into a rate of digital audio data corresponding to an accessed control signal according to the control signal; and the first half-band filter is configured to reduce a rate of digital audio data output by the multi-rate filter.