Fraunhofer-gesellschaft Zur Foerderung Der Angewandten Forschung E.v.

Other

Heinz Gerh&auml;user

Karlheinz Brandenburg

J&uuml;rgen Herre

Ernst Eberlein

Rainer Buchta

Bernhard Grill

Johannes Hilpert

Martin Dietz

Rainer Perthold

Oliver Kunz

J&uuml;rgen Zeller

Martin Sieler

Marco Breiling

Albert Heuberger

Wolfgang Alexander Schildbach

Bodo Teichmann

Christian Neubauer

Uwe Gbur

Hartmut Schott

Dietmar Schill

Roland Plankenb&uuml;hler

Jan Stoessel

A method for detecting a transient in a discrete-time audio signal is performed completely in the time domain and includes the step of segmenting the discrete-time audio signal so as to generate consecutive segments of the same length with unfiltered discrete-time audio signals xs(T&minus;1). The discrete-time audio signal in a current segment is subsequently filtered. Then either the energy of the filtered discrete-time audio signal in the current segment can be compared with the energy of the filtered discrete-time audio signal in a preceding segment or a current relationship between the energy of the filtered discrete-time audio signal in the current segment and the energy of the unfiltered discrete-time audio signal in the current segment can be formed and this current relationship compared with a preceding corresponding relationship. On the basis of the one and/or the other of these comparisons it is detected whether a transient is present in the discrete-time audio signal.

Method and device for detecting a transient in a discrete-time audio signal

In a method for signalling a noise substitution when coding an audio signal, the time-domain audio signal is first transformed into the frequency domain to obtain spectral values. The spectral values are subsequently grouped together to form groups of spectral values. On the basis of a detection establishing whether a group of spectral values is a noisy group or not, a codebook is allocated to a non-noisy or tonal group by means of a codebook number for redundancy coding of the same. If a group is noisy, an additional codebook number which does not refer to a codebook is allocated to it in order to signal that this group is noisy and therefore does not have to be redundancy coded. By signalling noise substitution by means of a Huffman codebook number for noisy groups of spectral values, which are e.g. sections made up of scale factor bands which do not have to be redundancy coded, an opportunity is provided to indicate the presence of a noise substitution in a scale factor band in the bit stream syntax of the MPEG-2 Advanced Audio Coding (AAC) Standard without having to interfere with the basic coding structure and without having to meddle with the structure of the existing bit stream syntax.

Method for signalling a noise substitution during audio signal coding

For reducing a peak value in single-carrier modulated or multi-carrier modulated digital transmitter signals, the peak value is first determined from a plurality of transmitter symbols contained in the digital signal. If a predetermined value is exceeded, a replacement symbol is generated which replaces a predetermined transmitter symbol. The replacement symbol is so chosen that it lowers the peak value and is convertible into the original transmitter symbol without error using a coding which was used to create the transmitter symbols.

Method for reducing peak values in single-carrier modulated or multi-carrier modulated digital transmitter signals

In a coding method and a coder for introducing a non-audible data signal into an audio signal, the audio signal is first transformed to a spectral range and the masking threshold of the audio signal is determined. A pseudo-noise signal and a data signal are provided and multiplied with each other so a to provide a frequency-spread data signal. The spread data signal is weighted with the masking threshold, and thereafter the audio signal and the weighted data signal are superimposed. In a method and a decoder for decoding a data signal introduced into an audio signal in non-audible manner, the audio signal is first sampled and thereafter the sampled audio signal is filtered in non-recursive manner. The filtered audio signal is subsequently compared to a threshold value so as to retrieve the data signal.

Coding process for inserting an inaudible data signal into an audio signal, decoding process, coder and decoder

A method for detecting a transient in a discrete-time audio signal is performed completely in the time domain and includes the step of segmenting the discrete-time audio signal as to generate consecutive segments of the same length with unfiltered discrete-time audio signals. The discrete-time audio signal in a current segment is filtered. Either the energy of the filtered discrete-time audio signal in the current segment is compared with the energy of the filtered discrete-time audio signal in a preceding segment or a current relationship between the energy of the filtered discrete-time audio signal in the current segment and the energy of the unfiltered discrete-time audio signal in the current segment is formed and this current relationship compared with a preceding corresponding relationship. Whether a transient is present in the discrete-time audio signal is detected using one and/or the other of these comparisons.

Method and device for detecting a transient in a discrete-time audiosignal

In a method for concealing errors in an audio data stream the occurrence of an error is detected in the audio data stream, audio data prior to the occurrence of the fault being intact audio data. Thereafter a spectral energy of a subgroup of the intact audio data is calculated. After forming a pattern for substitute data on the basis of the spectral energy calculated for the subgroup of the intact audio data, substitute data for erroneous or missing audio data which correspond to the subgroup are created on the basis of the pattern.

Method for masking defects in a stream of audio data

In a method for detecting a manipulation performed on digital information the erroneous determination of a manipulation during processing disturbances is prevented by subdividing the information into information subunits from each of which an identification is derived which is assigned to the respective information subunit. After processing the information subunits together with their identifications, potentially manipulated information subunits are determined on the basis of their identifications. This enables the whole of the information to be classified as manipulated if at least two information subunits which are correlated with respect to time or location have been determined to have been potentially manipulated.

Process for sensing manipulations of digital information

An apparatus for transmitting information comprises a bitstream source for providing a bitstream representing the information, a redundancy adding encoder for generating an encoded bitstream, which is arranged to output, for a first number of input bits, a second number of output bits, the second number of output bits having at least twice as many output bits as the first number of input bits, wherein the second number of output bits includes two portions of output bits, each portion of output bits individually allowing the retrieval of information represented by the first number of input bits, and the first portion of output bits being coded based on the bitstream in a different way with respect to the second portion of output bits. The apparatus further comprises a partitioner for partitioning the second number of output bits into the two portions of output bits and a transmitter for transmitting the output bits of the first portion via a first channel and the output bits of the second portion via a second channel, the second channel being spatially different from the first channel. An inventive receiving apparatus combines the signals received via the first and second channels and uses both channel signals for channel decoding by removing redundancy. Thus, the transmitting receiving system is suitable for providing time and/or space diversity and, in the optimal case, provides a C/N value which is greater than 4.3 dB with respect to a two-channel system comprising a duplicator in the transmitter and a channel-controlled switch in the receiver.

Apparatus and method for transmitting information and apparatus and method for receiving information

A circuit for predistorting a signal to be transmitted via a non-linear channel comprises an envelope detector for detecting an envelope of the signal. A quantization for forming quantized envelope values on the basis of the detected envelope is connected to the envelope detector as well as to a table for supplying complex predistortion coefficients which depend on the quantized envelope values and on a previously detected transmission function of the non-linear channel. An evaluation circuit separated from the non-linear channel and used for the complex evaluation of the signal to be transmitted via the non-linear channel with the complex predistortion coefficients permits the distortion introduced by the non-linear channel to be substantially compensated according to amount and phase.

Pre-distortion for a non-linear transmission path in the high frequency range

In a method for coding an audio signal digitized at a low sampling rate to obtain time domain audio samples. A frequency domain representation of the time domain audio samples is produced. The frequency domain representation includes successive frequency lines. These frequency lines are grouped into a plurality of scale factor bands. The successive frequency lines in a scale factor band are coded with the same scale factor. A plurality of regions is formed by grouping the scale factor bands, wherein successive scale factor bands form a region within which all the scale factors are coded with the same number of bits, which is determined according to the largest scale factor of the region. The scale factors assigned to scale factor bands within the highest region that includes the higher frequency successive frequency lines are set to zero. The frequency lines in the highest region are coded using the zero-valued scale factors that correspond to a multiplication factor of 1. The scale factors for the highest region, however, are not coded. Thus, the bits that would be required for coding these zero-valued scale factors are saved and can be used for a finer quantization of the rest of the spectrum. Additionally, this coding method when applied to ISO/IEC 13818-3 as a low sampling rate modification thereof only requires minimal changes with respect to this Standard.

Growing community of inventors

Waischenfeld, Germany

Heinz Gerhäuser

Growing community of inventors

Waischenfeld, Germany

Heinz Gerh&auml;user

Heinz Gerhäuser