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.

Diffuse or spatially large audio objects may be identified for special processing. A decorrelation process may be performed on audio signals corresponding to the large audio objects to produce decorrelated large audio object audio signals. These decorrelated large audio object audio signals may be associated with object locations, which may be stationary or time-varying locations. For example, the decorrelated large audio object audio signals may be rendered to virtual or actual speaker locations. The output of such a rendering process may be input to a scene simplification process. The decorrelation, associating and/or scene simplification processes may be performed prior to a process of encoding the audio data.

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.

An apparatus includes a receiver and an up-mixer. The receiver is configured to receive a bitstream that includes an encoded mid signal and encoded stereo parameter information. The encoded stereo parameter information represents a first value of a stereo parameter and a second value of the stereo parameter. The first value is associated with a first frequency range. The second value is associated with a second frequency range that is distinct from the first frequency range. The up-mixer is configured to perform an up-mix operation on a frequency-domain decoded mid signal generated from the encoded mid signal. A particular value based on the first value and the second value is applied to the frequency-domain decoded mid signal during the up-mix operation.

The present technology relates to a speaker array designed to be capable of achieving sufficiently high reproducibility at low cost, and a signal processing apparatus. The speaker array is formed with a plurality of higher order speakers and a plurality of general speakers. The type, the number, or the installation positions of the higher order speakers are determined in accordance with wavefront reproducibility in a second region located on the outer side of a first region that can be controlled by the general speakers. The present technology can be applied to a speaker array and a sound field forming apparatus.

A system can include a position sensor configured to output position data of a HWD. The system can include one or more processors configured to identify a first head angle of the HWD using the position sensor, generate an audio signal using the first head angle, identify a second head angle of the HWD using the position sensor, determine an angle error based at least on the first head angle and the second head angle, and apply at least one of a time difference or a level difference to the audio signal based at least on the angle error to adjust the audio signal. The system can include an audio output device configured to output the adjusted audio signal. By adjusting the audio signal using the angle error, the system can correct for long spatial update latencies and reduce the perceptual impact of such latencies for the user.

Method for live manipulation of signal flows via a controller, wherein the method comprises of feeding in a first signal flow and a further signal flow, each having X signal flow layers, wherein X is greater than 2. The method further comprises of separating the signal flow layers from each signal flow into a respective series of sub-signal flows, related to the signal flow, as according to a predetermined ratio, wherein each sub-signal flow has Y sub-signal flow layers, wherein Y is smaller than X. The method comprises of reading a desired ratio between the first signal flow and the further signal flow via a controller. The method comprises of merging corresponding sub-signal flows as according to the desired ratio in order to obtain a modified series of sub-signal flows. The method comprises of feeding out the modified series.

A device for processing audio signals, including: first and second input units providing first and second input signals of first and second audio tracks, a decomposition unit to decompose the first input audio signal to obtain a plurality of decomposed signals, a playback unit configured to start playback of a first output signal obtained from recombining at least a first decomposed signal at a first volume level with a second decomposed signal at a second volume level, such that the first output signal substantially equals the first input signal, and a transition unit for performing a transition between playback of the first output signal and playback of a second output signal obtained from the second input signal. The transition unit has a volume control section adapted for reducing the first and second volume levels according to first and second transition functions.

The present invention relates to a method and an apparatus for processing an audio signal, and more particularly, to a method and an apparatus for processing an audio signal, which synthesize an object signal and a channel signal and effectively perform binaural rendering of the synthesized signal.

To this end, provided are a method for processing an audio signal, which includes: receiving an input audio signal including a multi-channel signal; receiving truncated subband filter coefficients for filtering the input audio signal, the truncated subband filter coefficients being at least some of subband filter coefficients obtained from binaural room impulse response (BRIR) filter coefficients for binaural filtering of the input audio signal and the length of the truncated subband filter coefficients being determined based on filter order information obtained by at least partially using reverberation time information extracted from the corresponding subband filter coefficients; obtaining vector information indicating the BRIR filter coefficients corresponding to each channel of the input audio signal; and filtering each subband signal of the multi-channel signal by using the truncated subband filter coefficients corresponding to the relevant channel and subband based on the vector information and an apparatus for processing an audio signal by using the same.

A three-dimensional sound reproducing method includes: acquiring a multichannel audio signal; rendering signals to a channel to be reproduced according to channel information and a frequency of the multichannel audio signal; and mixing the rendered signals.