摘要:
The aim of the invention is to correct the level in a wave field synthesis system comprising a wave field synthesis module and a loudspeaker array for providing sound in a projection zone. Said aim is achieved by determining (100) a correction factor (104) based on a set-point amplitude state in a projection zone, said set-point amplitude state depending on a position of the virtual source or a type of virtual source while the real amplitude state in the projection zone depends on the component signals for the loudspeakers due to the virtual source. The determined correction factor (104) is fed to a device (106) which manipulates the audio signal associated with the virtual source before said audio signal is fed into the wave field synthesis module or manipulates the component signals for the individual loudspeakers due to the virtual source in order to reduce a deviation between a set-point amplitude state and a real amplitude state at one or several points in the projection zone such that level artifacts occurring as a result of the finite number of loudspeakers being used are at least decreased in a wave field synthesis system, resulting in a more pleasant sound experience for a listener.
摘要:
The invention relates to a wave field synthesis device for driving an array of loudspeakers with driver signals, whereby the loudspeakers are placed at different defined positions. A driver signal for a loudspeaker is based on an audio signal, which is assigned to a virtual source having a virtual position with regard to the loudspeaker array, and on the defined position of the loudspeaker. Relevant loudspeakers of the loudspeaker array are firstly determined based on the position of the virtual source, a predefined listener position, and on the defined positions of the loudspeakers so that artifacts caused by loudspeaker signals, which move opposite a direction of the virtual source to the predefined listener position, are reduced. A device (24) for feeding the driver signal components for the relevant loudspeakers for the virtual source to the relevant loudspeakers is connected downstream from a device (20) for calculating the driver signal components for the relevant loudspeakers and for a virtual source, whereby no driver signal components for the virtual source are fed to loudspeakers of the loudspeaker array that do not belong to the relevant loudspeakers. Artifacts in an area of the listener space are hereby suppressed due to the generation of a wave field so that only the useful wavefield is heard without artifacts in this area.
摘要:
The invention relates to a loudspeaker comprising a membrane, a first excitation device (14) for producing structure-borne noise in the membrane (12), and a second excitation device (16) which is different to the first and is used to displace the membrane (12) in a longitudinal oscillating movement perpendicularly to the membrane expansion direction. According to the invention, the problem of the too weak bass tone reproduction or of the quantity opposing the invisible integration or installation is solved by introducing a second excitation system which moves the membrane (12), or the plate used as a membrane, back and forth in a uniform manner, in addition to the bending oscillations of the structure-borne noise. Sound reproduction is thus rendered possible over the entire acoustic frequency range, without hindering the aim of invisible integration or installation.
摘要:
According to the inventive method for inserting information into an audio signal, a time multiplex method is combined with a code multiplex method in order to preprocess the information which is to be inserted into the audio signal. During a time multiplex method, a spreading is carried out (22, 24) with two different data sequences in order to be able to distinguish a first time slot from additional time slots. The code multiplex channels are added (26) and weighted (26, 28, 30, 32) while taking into account a psychoacoustic masking threshold of the audio signal, whereupon the weighted code multiplex signal is combined (34) with the audio signal. The time slot of the information channel is used while detecting the information that is inserted into the audio signal in order to synchronize the second information channel which had been spread with a data sequence differing from the data sequence for the other time slots. This results in the provision of a very reliable data transmission in a first information channel as well as the provision of a data transmission with a high transmission rate in the second transmission channel.
摘要:
Exemplary embodiments of the present invention provide a method for creating a database. The method comprises the steps "receiving environmental sounds", which comprise, for example, parasitic noise and "buffered environmental sounds for a rolling time window", such as 30 or 60 seconds, or preferably more than 5 seconds. Alternatively, the method could also comprise the step of "deriving a parameter set for the environmental sounds" and "buffering the parameter set for the rolling time window". The buffered environmental sounds or the buffered parameter set are generally designated as a recording. In addition, the method comprises the step "obtaining a signal", said signal identifying one signal class (e.g. parasitic noise) of a plurality of signal classes (parasitic noise and non-parasitic noise) in the environmental sounds. The third basic step is "storing, as a response to the signal, the buffered recordings" in a memory, such as an internal or external memory. These obtaining and storing steps are repeated in order to construct the database, which has a plurality of buffered recordings for the same signal class.
摘要:
The aim of the invention is to achieve a cleaner separation of a first audio signal in a first region of an area to be exposed to sound emitted by a plurality of loudspeakers. For this purpose, a calculating element calculates the version of the audio signals resulting from the spatially selective playback of the audio signals in this first region, calculates a masking threshold on the basis of the version of the audio signal that is to be separated from the one or more other audio signals in this region, and influences the output of the audio signals for the spatially selective playback to the outputs of the plurality of loudspeakers on the basis of a comparison of the masking threshold with the version of one or more other, i.e. interfering, audio signals.
摘要:
The invention relates to a device for synchronizing an audio signal with a film, which consists of individual images, whereby each individual image has an illuminated time code. The inventive device comprises a unit (10) for detecting the illuminated time code for the sequence of individual images in order to obtain a recorded sequence of time codes. A time code generator (12) is also provided, which is designed for generating a sequence of synthesis time codes starting from a starting value. A decoder (14) is additionally provided in order to decode a time code of the detected sequence of time codes whereby furnishing the starting value for the time code generator (12). A detected time code and a corresponding synthesis time code are compared (16) in order to, in the event a phase deviation that exceeds a deviation threshold value has been established, subsequently manipulate (20) the synthesis time code for this individual code whereby modifying it with regard to the duration thereof. This synthesis time code is then fed to an audio processing unit (24) that is designed for providing, in a time-controlled manner, and in response to a detection of the synthesis time code for an individual image, the sampled values of the audio signal that are assigned to said individual image. A flexible system is hereby achieved via which any number of audio playback units can be synchronized with the film. In addition, the audio playback units, for synchronization purposes, are furnished with predefined synthesis time codes or with manipulated synthesis time codes.
摘要:
Disclosed is a device for establishing a correlation between a test audio signal (270) playable at a variable speed and a reference audio signal (274) representing a digitally stored version of the test audio signal (270). Said device comprises an apparatus (210) for determining a measure of a playing speed of the test audio signal (270), an apparatus (230) for varying a test sampling rate at which the test audio signal (270) is sampled so as to generate a modified test audio signal (272) in accordance with the measure of the test playing speed or for varying a reference sampling rate of the digitally stored reference audio signal (274) so as to create a modified reference audio signal (276) in accordance with the measure of the test playing rate. The varying apparatus (230) is configured for varying the test sampling rate or the reference sampling rate such that a deviation between a test playing rate assigned to the test audio signal (270) or a reference playing speed allocated to the modified reference audio signal (276) is reduced or a deviation between a test playing speed assigned to the modified test audio signal (272) and a reference playing speed allocated to the reference audio signal (274), or a deviation between a test playing speed assigned to the modified test audio signal (274) and a reference playing speed allocated to a modified reference audio signal (276), is reduced. The inventive device further comprises an apparatus (250) for comparing the modified test audio signal (272) and the reference audio signal (274), or the test audio signal (270) and the modified reference audio signal (276), or the modified test audio signal (272) and the modified reference audio signal (276), in order to obtain a result of the correlation.