Exemplary embodiments provide encoding and decoding methods, and associated encoders and decoders, for encoding and decoding of an audio scene which is represented by one or more audio signals. The encoder generates a bit stream which comprises downmix signals and side information which includes individual matrix elements of a reconstruction matrix which enables reconstruction of the one or more audio signals in the decoder.

An apparatus for generating an output downmix representation from an input downmix representation, wherein at least a portion of the input downmix representation is in accordance with a first downmixing scheme, includes: an upmixer for upmixing at least the portion of the input downmix representation using an upmixing scheme corresponding to the first downmixing scheme to obtain at least one upmixed portion; and a downmixer for downmixing the at least one upmixed portion in accordance with a second downmixing scheme different from the first downmixing scheme.

Methods and systems for audio source separation in real-time are described. In an embodiment, the present disclosure describes reading and decoding an audio source into PCM samples, fragmenting Pulse Code Modulation (PCM) samples into fragments, transforming fragments into spectrograms, performing audio source separation using a training database that includes a training dictionary and non-negative matrix factorization (NMF) to generate a set of component signals, and streaming the component signals to a playback engine. In an embodiment, a semantic equalizer graphical user allows for fading of individual component signals.

Embodiments herein are primarily described in the context of a system, a method, and a non-transitory computer readable medium for producing a sound with enhanced spatial detectability and a crosstalk simulation. The audio processing system receives a left and right input channel of an audio input signal, and performs an audio processing to generate an output audio signal. The system generates left and right spatially enhanced signals by gain adjusting side subband components and mid subband components of the left and right input channels. The audio processing system generates left and right crosstalk channels such as by applying a filter and time delay to the left and right input channels, and mixes the spatially enhanced channels with the crosstalk channels. In some embodiments, the system includes high/low frequency enhancement channels and passthrough channels derived from the input channels, which can be mixed with the output audio signal.

An apparatus comprising: an input configured to receive from at least one co-operating apparatus at least one audio signal; an audio signal analyzer configured to analyze the at least one audio signal to determine at least one audio component position relative to the at least one co-operating apparatus recording position; and a processor configured to determine an position value based on the at least one cooperating recording position and the apparatus position, and further configured to apply the position value to the at least one audio component position, such that the at least one audio component position is substantially aligned with the apparatus position.

According to an example embodiment, a method for processing a multi-channel input audio signal representing a sound field into a multi-channel output audio signal representing said sound field in accordance with a predefined loudspeaker layout is provided, the method comprising the following for at least one frequency band: obtaining spatial audio parameters that are descriptive of spatial characteristics of said sound field; estimating a signal energy of the sound field represented by the multi-channel input audio signal; estimating, based on said signal energy and the obtained spatial audio parameters, respective output signal energies for channels of the multi-channel output audio signal according to said predefined loudspeaker layout; determining a maximum output energy as the largest of the output signal energies across channels of said multi-channel output audio signal; and deriving, on basis of said maximum output energy, a gain value for adjusting sound reproduction gain in at least one of said channels of the multi-channel output audio signal.

According to a sound image direction sense processing method and apparatus, a left-ear channel signal, a right-ear channel signal, and a centered channel signal that are of a sound source are obtained; whether a direction of the sound source is a front direction is determined according to the left-ear channel signal, the right-ear channel signal, and the centered channel signal; and when the direction of the sound source is the front direction, at least one type of the following processing: front direction enhancing processing or rear direction weakening processing is performed separately on the left-ear channel signal and the right-ear channel signal. Therefore, a difference between front direction sense and rear direction sense of a sound image may be enlarged, so that accuracy of determining a direction of a sound source may be improved.

Accurate modeling of acoustic reverberation can be essential to generating and providing a realistic virtual reality or augmented reality experience for a participant. In an example, a reverberation signal for playback using headphones can be provided. The reverberation signal can correspond to a virtual sound source signal originating at a specified location in a local listener environment. Providing the reverberation signal can include, among other things, using information about a reference impulse response from a reference environment and using characteristic information about reverberation decay in a local environment of the participant. Providing the reverberation signal can further include using information about a relationship between a volume of the reference environment and a volume of the local environment of the participant.

The present disclosure provides methods, devices and computer program products which provide less complex and more flexible control of the introduced decorrelation in an audio coding system. According to the disclosure, this is achieved by calculating and using two weighting factors, one for an approximated audio object and one for a decorrelated audio object, for introduction of decorrelation of audio objects in the audio coding system.

A processor can be associated with a memory for storing convolution data. A plurality of M filters from a corresponding plurality of M input channels to a selected one output channel can be provided, wherein each filter can be represented by a corresponding index, m. Each of the M filters can be partitioned into K respective filter partitions, wherein each respective filter partition can be represented by a corresponding index, k. A frequency-domain representation of each filter partition can be provided, wherein each frequency-domain representation of a filter partition comprises N frequency bins and a corresponding frequency-domain filter coefficient, wherein each respective frequency bin can be represented by a corresponding index, n. The memory can store such information in an arrangement suitable for the processor to concurrently receive sufficient information to concurrently convolve a frame of each input signal with the respective filters.