摘要:
An input audio signal having an input temporal envelope is converted into an output audio signal having an output temporal envelope. The input temporal envelope of the input audio signal is characterized. The input audio signal is processed to generate a processed audio signal, wherein the processing de-correlates the input audio signal. The processed audio signal is adjusted based on the characterized input temporal envelope to generate the output audio signal, wherein the output temporal envelope substantially matches the input temporal envelope.
摘要:
At an audio encoder, cue codes are generated for one or more audio channels, wherein an envelope cue code is generated by characterizing a temporal envelope in an audio channel. At an audio decoder, E transmitted audio channel(s) are decoded to generate C playback audio channels, where C>=Eo1. Received cue codes include an envelope cue code corresponding to a characterized temporal envelope of an audio channel corresponding to the transmitted channel(s). One or more transmitted channel(s) are upmixed to generate one or more upmixed channels. One or more playback channels are synthesized by applying the cue codes to the one or more upmixedchannels, wherein the envelope cue code is applied to an upmixed channel or a synthesized signal to adjust a temporal envelope of the synthesized signal based on the characterized temporal envelope such that the adjusted temporal envelope substantially matches the characterized temporal envelope.
摘要:
The invention relates to a method for characterising a signal representing an audio content. A measure is determined (12) for a tonality of the signal, whereupon a statement is made (16) about the audio content of the signal on the basis of the measure determined for the tonality of the signal. The measure for the tonality is derived from a quotient which has an average sum value in the numerator, of the spectral constituents of the signal which are exponentiated with a first power (x), and has an average sum value in the denominator, of spectral constituents which are exponientiated with a second power (y), the first and second powers differing from each other. The measure for the tonality of the signal for the content analysis is robust in relation to a signal distortion, e.g. by means of MP3-coding, and has a high correlation with the content of the signal analysed.
摘要:
An inventive method for introducing information into a data stream including data about spectral values representing a short-term spectrum of an audio signal first performs a processing of the data stream to obtain the spectral values of the short-term spectrum of the audio signal. Apart from that, the information to be introduced are combined with a spread sequence to obtain a spread information signal, whereupon a spectral representation of the spread information is generated which will then be weighted with an established psychoacoustic maskable noise energy to generate a weighted information signal, wherein the energy of the introduced information is substantially equal to or below the psychoacoustic masking threshold. The weighted information signal and the spectral values of the short-term spectrum of the audio signal will then be summed and afterwards processed again to obtain a processed data stream including both audio information and information to be introduced. By the fact that the information to be introduced are introduced into the data stream without changing to the time domain, the block rastering underlying the short-term spectrum will not be touched, so that introducing a watermark will not lead to tandem encoding effects.
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 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 scalable coder that comprises a first coding device (102), a decoding device (104) and a second coding device (114). Said coder is furthermore provided with a phase distorter (10a, 10b), for reducing a non-linear frequency-dependent phase distortion which is introduced by the first coding device (102) and/or the decoding device (104) and which causes an increased differential mode signal in a comparative device (110). A low-power differential signal is obtained which can code the second coding device (114) with less bits, that is with a higher bit efficiency.