摘要:
When an audio signal is coded, coded signals of inferior quality and bit-rate as well as coded signals of high quality and bit-rate are transmitted to a decoder. The low-bit-rate audio signal is coded and transmitted to the decoder first, before a further coded signal is transmitted to the decoder and either alone or together with the first coded signal produces a high-quality decoded signal within the decoder during decoding. This results first in a low-quality decoded signal in the decoder before decoding of the high-quality signal becomes possible.
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:
The device described has the following features: the device has a first input terminal to which is applied the input signal of the acoustic signal processing system to be monitored, a second input terminal to which is applied the output signal of the system, and a signal processor. The signal processor calculates, by correlation between the two signals present at the two input terminals, the signal delay time of the system to be monitored. The signal processor continuously produces the difference signal from the signal present for a given time interval at the first input terminal and the signal present later at the second input terminal after the signal delay time. The signal processor calculates the spectral composition of the signal present at the first input terminal during the given time interval and the corresponding difference signal. The signal processor calculates the threashold of hearing of a human ear from the spectral composition of the signal present at the first input terminal and compares it with the corresponding difference signal.
摘要:
The invention relates to a loudspeaker system comprising several sonic converters (2) which are fixed in or to the rear of sound passage openings (3) in a carrier plate. The loudspeaker system is characterized in that the carrier plate is embodied in the form of a wall plate (1) for the interiors or outer facades of buildings and in that the sonic converters (2) are integrated into the carrier plate. The inventive loudspeaker system makes it possible to acoustically irradiate an area without the need for esthetically troublesome loudspeakers.
摘要:
The invention relates to a method for generating a second data stream from a first data stream containing a first start block and a first user data block comprising user data. According to said method the first start block is extracted (116) from the first data stream. Next the second start block for the second data stream is generated (118), after which at least a part of the first start block, which part contains information making it possible to trace the origins of the user data, is integrated (120) into the second start block. Finally the second user data block containing the same user data is generated (122) so that a complete second data stream can be obtained. The method provided for by the invention allows for the device-specific encoding of user data, makes it possible to obtain a flexible device-specific copy for other devices of a user and, notably, provides complete references as to the origins of a copy, which in turn permits effective copyright protection.
摘要:
The invention relates to a method for generating an encoded user data stream which contains a start block and a block containing encoded user data. According to said method a user data code is generated for a user data coding algorithm for encoding user data (102). To obtain the block containing encoded user data of the user data stream, the user data are encoded (104) by means of the user data code and user data coding algorithm generated. A part of the user data stream is processed (106) so as to derive information which characterizes that part of the user data stream. This information is combined with the user data by means of an invertible logic operation (108) so that a base value is obtained. The base value is encoded by means of a code consisting of two different codes and using an asymmetrical encoding method (110), the two different codes being the public or private code for the asymmetrical encoding method. An output value is obtained which is an encoded version of the user data code and the output value is then entered into the start block so that the user data stream is completed (112). Unauthorized changes to the start block or the user data themselves result in automatic destruction of the user data.
摘要:
When coding and decoding stereophonic spectral values, both the intensity stereo process and a prediction process are used to achieve a high data compression. When an intensity stereo coding is active in a section composed of scaling factor bands (28), prediction for the right channel (R) is deactivated in this area, so that the prediction results are not used to build the coded stereophonic spectral values. The predictor of the right channel (R) is supplied with stereophonic spectral values for this channel which are in turn decoded by an intensity stereo process, so that prediction for the right channel (R) can continue to adapt itself.
摘要:
In order to measure the fidelity of stereophonic audio signals, a stereophonic signal is used as reference signal (X) and a signal to be tested (X') is created by processing the reference signal (X), for example by coding then decoding the reference signal (X). Both signals (X, X') are transformed into the frequency range in order to create spectral data that are representative of each partial band (i). Signal values (Gi; Gi') for each partial band (i) of both reference signal (X) and signal to be tested (X') are determined from the spectral data of the channels (L, R) of the reference signal or signal to be tested. Conclusions are made about the fidelity of the stereophonic audio signals processed or coded by using a determined processing or coding technique from the comparison of the characteristic values (Gi; Gi') of signals belonging to the same partial band (i).