Abstract:
The present application refers to an apparatus (200) for estimating an overall mixing time based on at least a first pair of room impulse responses, the apparatus comprising a processing element (305) configured to determine differences between energy profiles of a first room impulse response of the first pair of room impulse responses anda second room impulse response of the first pair of room impulse responses at a plurality of different sample times of the first pair of room impulse responses,set a sample time of the plurality of sample times as a mixing time for the first pair of room impulse responses at which the difference between the energy profiles of the first room impulse response and the second room impulse response of the first pair of room impulse responses is equal to or below a threshold value, anddetermine the overall mixing time based on the mixing time for the first pair of room impulse responses.The present applicationfurther refers to a corresponding method for estimating an overall mixing time. (Fig. 9)
Abstract:
An apparatus and a method for enhancing a spatial perception of an audio signal are provided creating increased interaural-level differences. To obtain this effect, two dipoles are used: one for producing a left audio signal and one for producing a right audio signal.
Abstract:
A system (100) and a method for evaluating an acoustic transfer function, wherein the acoustic transfer function is a transfer function from one acoustic source to a reproduction area (RA) sampled by a limited number of microphone modules (120).
Abstract:
A method for acoustic scene playback comprises: providing recording data comprising microphone signals of one or more microphone setups positioned within an acoustic scene and microphone metadata of the one or more microphone setups, wherein each of the one or more microphone setups comprises one or more microphones and has a recording spot which is a center position of the respective microphone setup; specifying a virtual listening position, wherein the virtual listening position is a position within the acoustic scene; assigning each microphone setup of the one or more microphone setups one or more Virtual Loudspeaker Objects, VLOs, wherein each VLO is an abstract sound output object within a virtual free field; generating an encoded data stream based on the recording data, the virtual listening position and VLO parameters of the VLOs assigned to the one or more microphone setups; and decoding the encoded data stream based on a playback setup, thereby generating a decoded data stream; and feeding the decoded data stream to a rendering device, thereby driving the rendering device to reproduce sound of the acoustic scene at the virtual listening position. A playback apparatus and a computer program for performing the method for acoustic scene playback are also described.
Abstract:
The invention relates to an audio compression system (100) for compressing an input audio signal, the audio compression system (100) comprising a digital filter (101) for filtering the input audio signal, the digital filter (101) comprising a frequency transfer function having a magnitude over frequency, the magnitude being formed by an equal loudness curve of a human ear to obtain a filtered audio signal, and a compressor (103) being configured to compress the input audio signal upon the basis of the filtered audio signal to obtain a compressed audio signal.
Abstract:
A device (100) and a method for producing a sound field are disclosed. The device (100) comprises a plurality of loudspeakers (101) arranged at a plurality of locations within a plane and a processing circuitry configured to drive the plurality of loudspeakers (101). A first subset of the plurality of loudspeakers (101) defines a first rhombus within the plane and a second subset of the plurality of loudspeakers (101) defines a second rhombus within the plane. The first rhombus is oriented substantially perpendicular to the second rhombus. Further subsets of the plurality of loudspeakers may define further rhombic sub-arrays. The audio device may be implemented as a soundbar or a sound panel. Thus, an audio device providing a richer sound experience is disclosed.
Abstract:
The invention relates to an audio signal processing apparatus (100) for processing an input audio (101) signal to be transmitted to a listener in such a way that the listener perceives the input audio signal (101) to come from a virtual target position defined by an azimuth angle and an elevation angle relative to the listener, the audio signal processing apparatus (100) comprising a memory (103) configured to store a set of pairs of predefined left ear and right ear transfer functions, which are predefined for a plurality of reference positions relative to the listener, wherein the plurality of reference positions lie in a a two-dimensional plane, a determiner(105) configured to determine a pair of left ear and right ear transfer functions on the basis of the set of predefined pairs of predefined left ear and right ear transfer functions for the azimuth angle and the elevation angle of the virtual target position and an adjustment filter (107) configured to filter the input audio signal (101) on the basis of the determined pair of left ear and right ear transfer functions and an adjustment function (109) configured to adjust a delay between the left ear transfer function and the right ear transfer function of the determined pair of left ear and right ear transfer functions and a frequency dependence of the left ear transfer function and the right ear transfer function of the determined pair of left ear and right ear transfer functions as a function of the azimuth angle and/or the elevation angle of the virtual target position in order to obtain a left ear output audio signal (111a) and a right ear output audio signal (111b).
Abstract:
The invention relates to an apparatus (100) for manipulating an input audio signal associated to a spatial audio source within a spatial audio scenario, wherein the spatial audio source has a certain distance to a listener within the spatial audio scenario, the apparatus (100) comprising an exciter (101) adapted to manipulate the input audio signal to obtain an output audio signal, and a controller (103) adapted to control parameters of the exciter (101) for manipulating the input audio signal upon the basis of the certain distance.
Abstract:
A method (900) for processing an audio signal includes: decomposing (901) an audio signal (602a, 602b) comprising spatial information into a set of audio signal components; and processing (902) a first subset (606a) of the set of audio signal components according to a first processing scheme (603) and processing a second subset (606) of the set of audio signal components according to a second processing scheme (609) different from the first processing scheme (603), wherein the first subset (606a) comprises audio signal components corresponding to at least one frontal signal source (M) and the second subset (606) comprises audio signal components corresponding to at least one ambient signal source (SL, SR); and wherein the second processing scheme (609) is based on crosstalk cancellation.
Abstract:
An apparatus and a method for compressing a set of N binaural room impulse responses, BRIR, wherein each channel of an N channel audio signal (I1, I2,..., IN) is convolved with the corresponding compressed set of N BRIR.