The invention refers to a system for handling digital content including an input interface, a calculator, and an output interface. The input interface receives digital content and includes a plurality of input channels. At least one input channel receives digital content from a sensor or a group of sensors belonging to a recording session. The calculator provides output digital content by adapting received digital content to a reproduction session in which the output digital content is to be reproduced. The output interface outputs the output digital content and includes a plurality of output channels, wherein at least one output channel outputs the output digital content to an actuator or a group of actuators belonging to the reproduction session. Further, the input interface, the calculator, and the output interface are connected with each other via a network. The input interface is configured to receive digital content via by Ni input channels, where the number Ni is based on a user interaction, and/or the output interface is configured to output the output digital content via by No output channels, where the number No is based on a user interaction. The invention further refers to a corresponding method.

A spatial sound generation device including a storage (106) and a controller (102) and connected to a plurality of speakers (116) is provided. In the spatial sound generation device, referring to information indicating a movable sounding body, the controller varies a transfer characteristic for each time in accordance with movement of the sounding body and applies an inverse filtering to calculate a plurality of input signals for the respective speakers from a sound source signal indicating a sound emitted by the sounding body. The inverse filtering outputs the input signals into the speakers to form a three-dimensional acoustic wave front under boundary surface control in accordance with a transfer characteristic for a space in which the plurality of speakers are arranged.

Provided is a signal processing device including a display control unit for causing a display to display an image corresponding to a specified place, a sound-collection-signal input unit for inputting a sound collection signal of a sound collection unit that collects a user sound produced with microphones surrounding the user, an acoustic-signal processing unit for performing a first acoustic-signal process for reproducing a sound field where the user sound is sensed as if the sound were echoing in the place on the signal input by the sound-collection-signal input unit, based on a first transfer function measured in the place to indicate how a sound emitted on a closed surface inside the place echoes in the place and then is transferred to the closed-surface side, and a sound-emission control unit for causing a sound based on the processed signal to be emitted from speakers surrounding the user.

Provided is a signal processing device including a display control unit for causing a display to display an image corresponding to a specified place, a sound-collection-signal input unit for inputting a sound collection signal of a sound collection unit that collects a user sound produced with microphones surrounding the user, an acoustic-signal processing unit for performing a first acoustic-signal process for reproducing a sound field where the user sound is sensed as if the sound were echoing in the place on the signal input by the sound-collection-signal input unit, based on a first transfer function measured in the place to indicate how a sound emitted on a closed surface inside the place echoes in the place and then is transferred to the closed-surface side, and a sound-emission control unit for causing a sound based on the processed signal to be emitted from speakers surrounding the user.

Higher Order Ambisonics represents three-dimensional sound independent of a specific loudspeaker set-up. However, transmission of an HOA representation results in a very high bit rate. Therefore, compression with a fixed number of channels is used, in which directional and ambient signal components are processed differently. The ambient HOA component is represented by a minimum number of HOA coefficient sequences. The remaining channels contain either directional signals or additional coefficient sequences of the ambient HOA component, depending on what will result in optimum perceptual quality. This processing can change on a frame-by-frame basis.

The following coding scenario is addressed: A number of audio source signals need to be transmitted or stored for the purpose of mixing wave field synthesis, multi-channel surround, or stereo signals after decoding the source signals. The proposed technique offers significant coding gain when jointly coding the source signals, compared to separately coding them, even when no redundancy is present between the source signals. This is possible by considering statistical properties of the source signals, the properties of mixing techniques, and spatial hearing. The sum of the source signals is transmitted plus the statistical properties of the source signals, which mostly determine the perceptually important spatial cues of the final mixed audio channels. Source signals are recovered at the receiver such that their statistical properties approximate the corresponding properties of the original source signals. Subjective evaluations indicate that high audio quality is achieved by the proposed scheme.

Techniques for simulating a microphone array and generating synthetic audio data to analyze the microphone array geometry. This reduces the development cost of new microphone arrays by enabling an evaluation of performance metrics (False Rejection Rate (FRR), Word Error Rate (WER), etc.) without building device hardware or collecting data. To generate the synthetic audio data, the system performs acoustic modeling to determine a room impulse response associated with a prototype device (e.g., potential microphone array) in a room. The acoustic modeling is based on two parametersa device response (information about acoustics and geometry of the prototype device) and a room response (information about acoustics and geometry of the room). The device response can be simulated based on the microphone array geometry, and the room response can be determined using a specialized microphone and a plane wave decomposition algorithm.

An audio processing apparatus comprises a receiver (705) which receives audio data including audio components and render configuration data including audio transducer position data for a set of audio transducers (703). A renderer (707) generating audio transducer signals for the set of audio transducers from the audio data. The renderer (7010) is capable of rendering audio components in accordance with a plurality of rendering modes. A render controller (709) selects the rendering modes for the renderer (707) from the plurality of rendering modes based on the audio transducer position data. The renderer (707) can employ different rendering modes for different subsets of the set of audio transducers the render controller (709) can independently select rendering modes for each of the different subsets of the set of audio transducers (703). The render controller (709) can select the rendering mode for a first audio transducer of the set of audio transducers (703) in response to a position of the first audio transducer relative to a predetermined position for the audio transducer. The approach may provide improved adaptation, e.g. to scenarios where most speakers are at desired positions whereas a subset deviate from the desired position(s).

A sound processing apparatus includes a signal processing device that individually performs first localization setting (e.g., sound volume panning) for setting localization of an input sound signal based on a value of a first parameter and second localization setting (e.g., delay panning) for setting localization of the input sound signal based on a value of a second parameter. In response to an adjustment by an operation device of the value of one of the first and second parameters, a control device automatically changes the value of the other of the first and second parameters. In this way, sound image localization based on the first localization setting and sound image localization based on the second localization setting are automatically controlled in an interlocked relation to each other. At least one of the sound signals localized based on the first and/or second localization setting is output to an output destination.

A rendering system including a plurality of loudspeakers, at least one microphone and a signal processing unit. The signal processing unit is configured to determine at least some components of a loudspeaker-enclosure-microphone transfer function matrix estimate describing acoustic paths between the plurality of loudspeakers and the at least one microphone using a rendering filters transfer function matrix using which a number of virtual sources is reproduced with the plurality of loudspeakers.