摘要:
The invention relates to a device (10) for producing an encoded data stream which represents an audio and/or video signal. Said device comprises an encoder (16) for encoding an input signal (12) to produce a data stream of a defined data stream syntax as the output signal. Said device further comprises an encryption device (18) which is coupled to the encoder (16) to influence encoding-related data (20a) and/or the output signal (20b) of the encoder in an unequivocally reversible manner on the basis of a code in such a manner that the produced encoded data stream contains useful information that differs from the useful information of a data stream that would be produced by the device without the presence of the encryption device and that the produced encoded data stream has the defined data stream syntax. The invention thus provides a flexible data stream encryption according to which the degree of encryption can be freely selected in such a manner that the user of a decoder who does not possess the code still has a rough idea of the audio and/or video signal that might cause him/her to buy the code to hear or view the audio and/or video signal in its full quality. The encoder-specific encryption and decryption concept can be implemented into already existing encoders/decoders with little effort.
摘要:
The invention relates to a method for embedding a watermark in an audio signal, according to which a spectral representation of the audio signal and a spectral representation of the watermark signal are first determined (14, 16). The spectral representation of the watermark signal is then processed, based on a psychoacoustic masking threshold (24) of the audio signal (22). The processed watermark signal is combined with the audio signal (18), to obtain an audio signal furnished with a watermark. The spectral representation of the watermark signal is processed iteratively in the following way: a predetermined watermark initial value (26) is first selected; the interference that has been introduced into the spectral representation by the predetermined watermark initial value after a quantization of the spectral representation of the audio signal, is determined (28). If the interference introduced by the watermark initial value is greater than the predetermined interference threshold (32), the watermark initial value is then modified until said interference introduced into the spectral representation of the audio signal by a modified watermark initial value after quantization, is less than or equal to the predetermined interference threshold. The modified watermark initial value that is obtained at the end of the iteration is used as a processed watermark signal to be combined with the audio signal. It is therefore no longer possible to weaken or remove a watermark by quantization. The invention provides complete control over the energy of the watermark, in order to ensure that a watermark is embedded into an audio signal with either the best possible degree of detectability or the best possible audio quality.
摘要:
The invention relates to a method for characterising a signal representing an audio content. A quantity is determined (12) for a tonality of the signal, whereupon information relating to the audio content of the signal is obtained (16) on the basis of the quantity for the tonality of the signal. Said quantity for the tonality of the signal, used to analyse the content, is stable in relation to a signal distortion, e.g. resulting from MP3-coding, and has a high correlation with the content of the signal examined.
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 invention relates to a method for coding or de-coding an audio signal combining the advantages of TNS processing and noise substitution. A time discrete audio signal is initially transformed in a frequency range in order to obtain spectral value of the temporal audio signal. A prediction of the spectral values in relation to frequency is subsequently made in order to enable spectral residual values. Areas within the spectral values encompassing spectral values with noise properties are detected . The spectral residual values are noise substituted in the noise areas, whereupon data relating to the noise areas and noise substitution are incorporated into side information pertaining to a coded audio signal.
摘要:
A process for transmitting and/or storing digital signals from several channels is particularly suitable for transmitting the five channels of a 3/2 stereophonic system, as well as for transmitting two stereo channels and three additional commentary channels. This process allows for example television programmes to be broadcast together with multilingual audio signals. This process is characterised in that only a bit rate of 384 kbit/s is required for transmission, thanks to the reduction of the data to be transmitted. In order to reduce the data, K input channels are reproduced segment by segment on N « K virtual spectral data channels, the spectral data channels are quantified, coded and transmitted, taking into account the laws of psychoacoustics, and K output channels are reproduced from the transmitted bit flow by means of a list transmitted therewith from the N « K spectral data channels.
摘要:
An audio post-processor (100) for post-processing an audio signal (102) having a time-variable high frequency gain information (104) as side information comprises: a band extractor (110) for extracting a high frequency band (112) of the audio signal (102) and a low frequency band (114) of the audio signal (102); a high band processor (120) for performing a time-variable modification of the high frequency band (112) in accordance with the time-variable high frequency gain information (104) to obtain a processed high frequency band (122); and a combiner (130) for combining the processed high frequency band (122) and the low frequency band (114). Furthermore, a pre-processor is illustrated.
摘要:
An apparatus for providing one or more adjusted parameters for a provision of an upmix signal representation on the basis of a downmix signal representation and a parametric side information associated with the downmix signal representation comprises a parameter adjuster. The parameter adjuster is configured to receive one or more parameters and to provide, on the basis thereof, one or more adjusted parameters. The parameter adjuster is configured to provide the one or more adjusted parameters in dependence on an average value of a plurality of parameter values, such that a distortion of the upmix signal representation caused by the use of non-optimal parameters is reduced at least for parameters deviating from optimal parameters by more than a predetermined deviation.