Abstract: A method for the artificial extension of the bandwidth of speech signals involves: a) Provision of a wideband input speech signal (swbi(k)); b) Determination of the signal components (seb(k)) of the wideband input speech signal (swbi(k)) required for the bandwidth extension from an extension band from the wideband input speech signal (swbi(k)); c) Determination of the temporal envelopes of the signal components (seb(k)) determined for the bandwidth extension; d) Determination of the spectral envelopes of the signal components (seb(k)) determined for bandwidth extension; e) Encoding of the information for the temporal envelopes and the spectral envelopes, and provision of the encoded information by carrying out the extension of the bandwidth; f) Decoding of the encoded information and generation of the temporal envelopes and the spectral envelopes from the encoded information for the production of a bandwidth-extended output speech signal (swbo(k)).

Abstract: In a method for detecting a leak in a double pipe, a medium located in an intermediate chamber between the interior and exterior pipe is moved toward a first end of the pipe, whereupon the medium flows in from the second end. On the first end the medium is examined for any leakage of a characteristic material, upon the detection of which a leakage signal is generated and a location of the leakage is calculated based on the transport time of the characteristic material from the leakage to the first end and on the mass flow of the medium. A corresponding device contains a conveyor unit for moving the medium through the intermediate chamber. A material sensor is disposed at the first end for examining the medium for the characteristic material. A control and analysis unit is provided for generating a leakage signal and calculating the location of the leakage.

Abstract: Representations of spatial audio scenes using higher-order Ambisonics HOA technology typically require a large number of coefficients per time instant. This data rate is too high for most practical applications that require real-time transmission of audio signals. According to the invention, the compression is carried out in spatial domain instead of HOA domain. The (N+1)2 input HOA coefficients are transformed into (N+1)2 equivalent signals in spatial domain, and the resulting (N+1)2 time-domain signals are input to a bank of parallel perceptual codecs. At decoder side, the individual spatial-domain signals are decoded, and the spatial-domain coefficients are transformed back into HOA domain in order to recover the original HOA representation.

Abstract: According to the invention, an excitation signal is generated as a result of sampled excitation values in order to excite an audio synthesis filter, the generated sampled excitation values being continuously stored in an adaptive codebook. A noise generator is provided which continuously generates random sampled values. A sequence of the stored sampled excitation values is selected from the adaptive codebook based on a fed audio fundamental frequency parameter by means of which a time gap between the sequence that is to be selected and the actual time reference is predefined. The excitation signal is generated by mixing the selected sequence with a random sequence encompassing actual random sampled valued of the noise generator.

Abstract: In frame-based bit stream formats the data required for decoding a current frame are usually stored within the data section for that frame. One exception is the mp3 bit stream where data for a current frame is stored in previous frames. If the decoder did not receive the required previous frame, decoding of the current mp3 frame is skipped. The invention can be applied for such bit streams, in an archival mode, a streaming mode and a sample-exact cutting of an archival mode. In the streaming and cutting modes, new headers are established. The number of frames required for initialising the decoder status is signalised in the header, as well as a consistency check value in the streaming mode. These frames are used for decoder initialisation but not for decoding samples or coefficients. For a sample-exact cutting, for the frame at which the cut shall occur, the number of samples or coefficients to be muted is also indicated in the header.

Abstract: Multimedia application data formats often use data packets and may contain packetized hierarchical layers with a base layer (BL) and one or more enhancement layers (EL). Packets that comprise variable length coded data typically provide padding bits, which can be used for adding checksum information. The checksum information refers to both, the decoded base layer and decoded enhancement layer packet data, and is used for detecting synchronization loss between the layers. Though the packets provide different individual amounts of padding bits, this variable amount of bits per packet is sufficient for adding check-sum information for synchronization related purposes. This allows e.g. out-of-sync detection even if the enhancement layer encoding provides no overhead information for synchronization, in particular no packet headers and no packet synchronization words.

Abstract: An audio signal may have a BL and an EL, wherein the EL represents additional information for enhancing the quality of the BL audio content. Decoding of such dual-layer signals usually comprises partial decoding of the BL data, wherein frequency bins of the BL are restored, mapping the restored frequency bins to the MDCT domain, adding them to the decoded EL and performing inverse Integer MDCT. A low-complexity method for decoding comprises reverse mapping of the decoded EL data, adding the reverse mapped EL data to the partially decoded BL data and filtering the sum, using the inverse BL filter bank.

Abstract: Filter banks may have different structures and different individual output signal domains. Often a translation between different filter bank domains is desirable. Usually, mapping matrices are used that, however, vary over frequency. This requires a significant amount of lookup tables. A method for transforming first data frames of a first filter bank domain to second data frames of a different second filter bank domain, comprises steps of transcoding sub-bands of the first filter bank domain into sub-bands of an intermediate domain that corresponds to said second filter bank domain but has warped phase, and transcoding the sub-bands of the intermediate domain to sub-bands of the second filter bank domain, wherein a phase correction is performed on the sub-bands of the intermediate domain.

Abstract: Integer MDCT is used in lossless audio coding. However, an integer-reversible transformation is split into consecutive lifting steps, each of which introduces considerable rounding errors to the signal. This problem gets more significant the lower the level of the useful signal to be encoded. Without noise shaping the rounding error noise will impact all frequency bins of the transformed signal equally. This is a particular problem for frequency bins in which the actual signal level is low. The invention limits the impact of rounding error noise coming with, or resulting from, each lifting step in the integer-reversible transformation on the data rate of a lossless codec. The filter coefficients of an adaptive noise shaping filter for transform coefficients are adapted in individual lifting steps according to the current time domain signal characteristics.

Abstract: A collector line for leakage monitoring and leakage location at a plant includes a carrier tube having a wall with openings formed therein, which are sealed with silicone rubber. This makes it possible to use the collector line during low ambient temperatures. A device and a method for leakage monitoring and leakage location and a pipeline having a collector line, are also provided.

Abstract: In a method for detecting a leak in a double pipe, a medium located in an intermediate chamber between the interior and exterior pipe is moved toward a first end of the pipe, whereupon the medium flows in from the second end. On the first end the medium is examined for any leakage of a characteristic material, upon the detection of which a leakage signal is generated and a location of the leakage is calculated based on the transport time of the characteristic material from the leakage to the first end and on the mass flow of the medium. A corresponding device contains a conveyor unit for moving the medium through the intermediate chamber. A material sensor is disposed at the first end for examining the medium for the characteristic material. A control and analysis unit is provided for generating a leakage signal and calculating the location of the leakage.

Abstract: A method and a device for detecting a pulse-type mechanical effect at a system component, include continuously recording an operating noise in the system component with a sensor disposed at the system component and converting the operating noise with the sensor into a measured signal subjected to a mathematical transformation. A sliding mean is calculated by determining quantiles from determined transformations and an evaluation function, which displays an application of the pulse-type mechanical effect on the system component, is derived from the sliding mean.

Abstract: The invention is related to lossless encoding of a source signal, using a lossy encoded data stream and a lossless extension data stream which together form a lossless encoded data stream for said source signal, whereby lossless audio compression means audio coding with bit-exact reproduction of the original PCM samples at decoder output. The lossy encoding/decoding may be an mp3 coding/decoding. The invention uses an integer MDCT and frequency domain de-correlation and time domain de-correlation for the residual signal of the base-layer lossy audio codec. The exploitation of side information from the lossy base-layer codec allows for reduction of redundancies in the gross bit stream, thus improving the coding efficiency of the lossy based lossless codec.

Abstract: The present invention provides method and device for transcoding between audio coding formats with different time-frequency analysis domains, as used for example by MPEG-AAC and mp3, particularly for facilitated and faster transcoding between such audio signals. A method for transcoding a framed audio signal from a first parameter domain into a second parameter domain comprises linearly transforming two or more parameters of the first parameter domain to at least one parameter of the second parameter domain, wherein the two or more parameters of the first parameter domain come from different frames of the audio signal in the first parameter domain. The linear transformation can be described as a matrix and implemented as a look-up table.

Abstract: Lossless compression algorithms can only exploit redundancies of the original audio signal to reduce the data rate, but not irrelevancies as identified by psycho-acoustics. Lossless audio coding schemes apply a filter or transform for decorrelation and then encode the transformed signal. The encoded bit stream comprises the parameters of the transform or filter, and the lossless representation of the transformed signal. However, in case of lossy based lossless coding the additional amount of information exceeds the amount of data for the base layer by a multiple of the base layer data amount. Therefore the additional data cannot be packed completely into the base layer data stream e.g. as ancillary data. The at least two data streams resulting from the combination of lossy coding format with a lossless coding extension are the base layer containing the lossy coding information and the enhancement data stream for rebuilding the mathematically lossless original input signal.

Abstract: In lossy based lossless coding a PCM audio signal passes through a lossy encoder to a lossy decoder. The lossy encoder provides a lossy bit stream. The lossy decoder also provides side information that is used to control the coefficients of a prediction filter that de-correlates the difference signal between the PCM signal and the lossy decoder output. The de-correlated difference signal is lossless encoded, providing an extension bit stream. Instead of, or in addition to, de-correlating in the time domain, a de-correlation in the frequency domain using spectral whitening can be performed. The lossy encoded bit stream together with the lossless encoded extension bit stream form a lossless encoded bitstream. The invention facilitates enhancing a lossy perceptual audio encoding/decoding by an extension that enables mathematically exact reproduction of the original waveform, and provides additional data for reconstructing at decoder site an intermediate-quality audio signal.

Abstract: In lossy based lossless coding a PCM audio signal passes through a lossy encoder to a lossy decoder. The lossy encoder provides a lossy bit stream. The difference signal between the PCM signal and the lossy decoder output is lossless encoded, providing an extension bit stream. The invention facilitates enhancing a lossy perceptual audio encoding/decoding by an extension that enables mathematically exact reproduction of the original waveform using enhanced de-correlation, and provides additional data for reconstructing at decoder site an intermediate-quality audio signal. The lossless extension can be used to extend the widely used mp3 encoding/decoding to lossless encoding/decoding and superior quality mp3 encoding/de-coding.

Abstract: A method and device detect a position of a pulsed mechanical effect on a plant component. An operating noise in the plant component is continuously recorded by sensors which are arranged on the plant component and converted by the above into a measured signal. The measured signals of the sensors undergo a transformation in a first time window. A first evaluation function is derived from a plurality of first transformations determined in this manner, the evaluation functions display an appearance of the pulsed mechanical effect. Accordingly, when detecting an effect following the second shorter time window having the same algorithms, second transformed and respectively, second evaluation functions are derived, from which, respectively, one time point is determined. Wherein the sound signal produced by the effect impacts upon the sensor. From there, running time differences produced between the sensors can be exactly reconstructed on the position of the effect.

Abstract: At the time of encoding audio content, the finally required data rate for delivery to the customer may be unknown. A data format is disclosed that is optimized for serving as Intermediate Format for efficient and fast recoding, to obtain one or more standard complying lossy encoded data streams with flexible data rates. Encoding can be performed in two steps that are inter-coordinated for cooperating, but may be locally and/or temporally separate. Between the partial encoders encoding parameters and/or auxiliary data are transmitted in a separate parameter enhancement layer, which complements a lossy data stream and can be used by the second encoder or transcoder for fast and computationally efficient implementation of the second encoding step. An additional lossless enhancement layer allows lossless reconstruction.

Abstract: A method and a device for detecting a pulse-type mechanical effect at a system component, include continuously recording an operating noise in the system component with a sensor disposed at the system component and converting the operating noise with the sensor into a measured signal subjected to a mathematical transformation. A sliding mean is calculated by determining quantiles from determined transformations and an evaluation function, which displays an application of the pulse-type mechanical effect on the system component, is derived from the sliding mean.