A method of converting an audio signal into signals for a number of loudspeaker transducers, where the audio signal is divided up into audio sub signals each representing a particular frequency interval, and where the signal for each loudspeaker transducer comprises a portion of each audio sub signal which varies over time.

In various embodiments, a method includes receiving a first audio channel signal; causing a first driver and a second driver of an audio output device to output a first frequency range of the first audio channel signal with a first power output; causing the first driver to output a second frequency range of the first audio channel signal with a second power output; and causing the second driver to output the second frequency range of the first audio channel signal with a third power output that is higher than the second power output, wherein the second frequency range is higher than the first frequency range.

Example techniques facilitate calibration of a playback device. An example implementation involves a computing device capturing, via a microphone, data representing multiple iterations of a calibration sound as played by a playback device. The computing device identifies multiple sections within the captured data. Two or more sections represent respective iterations of the calibration sound as played by the playback device. Based on the multiple identified sections, the computing device determines a frequency response of the playback device, the frequency response of the playback device representing audio output by the playback device and acoustic characteristics of an environment around the playback device. Based on the frequency response of the playback device and a target frequency response, the computing device determines one or more parameters of an audio processing algorithm and sends, to the playback device, the one or more parameters of the audio processing algorithm.

The present technique pertains to a signal processing apparatus, a method, and a program that enable even a low-cost apparatus to perform high-quality audio reproduction. The signal processing apparatus includes an obtainment unit that obtains a first audio signal, first band expansion information for band expansion of the first audio signal, and second band expansion information for band expansion of a second audio signal obtained by performing predetermined signal processing on the first audio signal, a selection unit that selects which of the first band expansion information and the second band expansion information to perform band expansion on the basis of, and a band expansion unit that, on the basis of the selected first band expansion information or second band expansion information and the first audio signal or the second audio signal performs band expansion, and generates a third audio signal. The present technique can be applied to a signal processing apparatus.

A system includes one or more computing devices that encode spatial perceptual cues into a monaural channel to generate a plurality of output channels. A computing device determines a target amplitude response for the mid and side channels of the plurality of output channels, defining a spatial perceptual associated with one or more frequency-dependent phase shifts. The computing device determines a transfer function of a single-input, multi-output allpass filter based on the target amplitude response and determines coefficients of the allpass filter based on the transfer function, and processes the monaural channel with the coefficients of the allpass filter to generate the plurality of channels having the encoded spatial perceptual cues. The allpass filter is configured to be colorless with respect to the individual output channels, allowing for the placement of spatial cues into the audio stream to be decoupled from the overall coloration of the audio.

An apparatus for providing a contrast mode and a front optimized mode for audio in a vehicle is provided. An audio controller is programmed to transmit first audio content in a first zone seating area and to transmit second audio content in a second zone seating area. The audio controller receives a first indication to transmit the first audio content in the first zone seating area and the second audio content in the second zone seating area in the contrast mode to provide an equal listening experience. The audio controller receives a second indication to transmit the first audio content in the first zone seating area and the second audio content in the second zone seating area in the front optimized mode to increase a quality of sound in the first zone seating area and to decrease a quality of sound in the second zone seating area.

Provided are a method and apparatus for localizing a multichannel sound signal. The method includes: obtaining a multichannel sound signal to which sense of elevation is applied by applying a first filter to an input sound signal; determining at least one frequency range of a dynamic cue according to change of a head-related transfer function (HRTF) indicating information regarding paths from a spatial location of an actual speaker to ears of an audience; and applying a second filter to at least one sound signal, corresponding to the determined at least one frequency range, of at least one channel in the multichannel sound signal to change the at least one sound signal so as to remove or to reduce the dynamic cue when the multichannel sound signal is output.

Improved methods and/or apparatus for decoding an encoded audio signal in soundfield format for L loudspeakers. The method and/or apparatus can render an Ambisonics format audio signal to 2D loudspeaker setup(s) based on a rendering matrix. The rendering matrix has elements based on loudspeaker positions and wherein the rendering matrix is determined based on weighting at least an element of a first matrix with a weighting factor $g=\frac{1}{\sqrt{L}}.$
The first matrix is determined based on positions of the L loudspeakers and at least a virtual position of at least a virtual loudspeaker that is added to the positions of the L loudspeakers.

Methods and systems for designing binaural room impulse responses (BRIRs) for use in headphone virtualizers, and methods and systems for generating a binaural signal in response to a set of channels of a multi-channel audio signal, including by applying a BRIR to each channel of the set, thereby generating filtered signals, and combining the filtered signals to generate the binaural signal, where each BRIR has been designed in accordance with an embodiment of the design method. Other aspects are audio processing units configured to perform any embodiment of the inventive method. In accordance with some embodiments, BRIR design is formulated as a numerical optimization problem based on a simulation model (which generates candidate BRIRs) and at least one objective function (which evaluates each candidate BRIR), and includes identification of a best one of the candidate BRIRs as indicated by performance metrics determined for the candidate BRIRs by each objective function.

A method for representing a second presentation of audio channels or objects as a data stream, the method comprising the steps of: (a) providing a set of base signals, the base signals representing a first presentation of the audio channels or objects; (b) providing a set of transformation parameters, the transformation parameters intended to transform the first presentation into the second presentation; the transformation parameters further being specified for at least two frequency bands and including a set of multi-tap convolution matrix parameters for at least one of the frequency bands.