100 results about "Spectral representation" patented technology

A time-discrete audio signal is processed to provide a quantization block with quantized spectral values. Furthermore, an integer spectral representation is generated from the time-discrete audio signal using an integer transform algorithm. The quantization block having been generated using a psychoacoustic model is inversely quantized and rounded to then form a difference between the integer spectral values and the inversely quantized rounded spectral values. The quantization block alone provides a lossy psychoacoustically coded/decoded audio signal after the decoding, whereas the quantization block, together with the combination block, provides a lossless or almost lossless coded and again decoded audio signal in the decoding. By generating the differential signal in the frequency domain, a simpler coder/decoder structure results.

An audio signal synthesizer generates a synthesis audio signal having a first frequency band and a second synthesized frequency band derived from the first frequency band and comprises a patch generator, a spectral converter, a raw signal processor and a combiner. The patch generator performs at least two different patching algorithms, each patching algorithm generating a raw signal. The patch generator is adapted to select one of the at least two different patching algorithms in response to a control information. The spectral converter converts the raw signal into a raw signal spectral representation. The raw signal processor processes the raw signal spectral representation in response to spectral domain spectral band replication parameters to obtain an adjusted raw signal spectral representation.

A noise filler for providing a noise-filled spectral representation of an audio signal on the basis of an input spectral representation of the audio signal has a spectral region identifier configured to identify spectral regions of the input spectral representation spaced from non-zero spectral regions of the input spectral representation by at least one intermediate spectral region, to obtain identified spectral regions, and a noise inserter configured to selectively introduce noise into the identified spectral regions to obtain the noise-filled spectral representation of the audio signal. A noise filling parameter calculator for providing a noise filling parameter on the basis of a quantized spectral representation of an audio signal has a spectral region identifier, as mentioned above, and a noise value calculator configured to selectively consider quantization errors of the identified spectral regions for a calculation of the noise filling parameter. Accordingly, an encoded audio signal representation representing the audio signal can be obtained.

A filter bank device for generating a complex spectral representation of a discrete-time signal includes a generator for generating a block-wise real spectral representation, which, for example, implements an MDCT, to obtain temporally successive blocks of real spectral coefficients. The output values of this spectral conversion device are fed to a post-processor for post-processing the block-wise real spectral representation to obtain an approximated complex spectral representation having successive blocks, each block having a set of complex approximated spectral coefficients, wherein a complex approximated spectral coefficient can be represented by a first partial spectral coefficient and by a second partial spectral coefficient, wherein at least one of the first and second partial spectral coefficients is determined by combining at least two real spectral coefficients. A good approximation for a complex spectral representation of the discrete-time signal is obtained by combining two real spectral coefficients, preferably by a weighted linear combination, wherein additionally more degrees of freedom for optimizing the entire system are available.

The finding of the present invention is that the melody extraction or automatic transcription may be implemented clearly more stable and if applicable even less expensive when the assumption is considered sufficiently that the main melody is the portion of a piece of music which man perceives the loudest and the most precise. Regarding this, according to the present invention the time/spectral representation or the spectrogram of an interesting audio signal is scaled using the curves of equal volume reflecting human volume perception in order to determine the melody of the audio signal on the basis of the resulting perception-related time/spectral representation.

Prior to embedding a watermark in an audio signal, a spectral representation of the audio signal and a spectral representation of the watermark signal are determined. The spectral representation of the watermark signal is then processed on the basis of a psychoacoustic masking threshold of the audio signal. The processed watermark signal is combined with the audio signal to obtain an audio signal bearing a watermark. The spectral representation of the watermark signal is processed iteratively as follows: first a predetermined watermark initial value is selected, then the interference introduced into the spectral representation of the audio signal after a quantization of the spectral representation of the audio signal is determined and then, if the interference introduced by the watermark initial value exceeds the predetermined interference threshold, the watermark initial value is modified progressively until the resulting interference introduced into the spectral representation of the audio signal after quantization is less than or equal to the predetermined interference threshold. The modified watermark initial value at the end of the iteration is used as the processed watermark signal to be combined with the audio signal. As a result it is no longer possible for a watermark to be quantized out. Instead, full control over the energy of the watermark is achieved. A watermark can therefore be embedded in an audio signal to provide either the best possible degree of watermark detectability or the best possible audio quality.

There is disclosed, in one aspect, a signal processing device (100) including: processing means (108) for generating a spectral representation of an input sound signal; frequency transposition means (110) for transposing at least part of the input signal's spectral representation to a transposed output frequency, said frequency transposition means (110) being configured to process the portion of the input signal spectral representation such that a phase relationship that existed in the input signal's spectral representation is substantially maintained in the transposed portion of the spectral representation; and synthesis means (112) for generating an output signal including the transposed portion of the input signal. Associated methods of processing a received sound signal are also disclosed.

The present invention provides an apparatus, method and tangible medium storing instructions for determining tonality of an input audio signal, for selection of corresponding masked thresholds for use in perceptual audio coding. In the various embodiments, the input audio signal is sampled and transformed using a compressed spectral operation to form a compressed spectral representation, such as a cepstral representation. A peak magnitude and an average magnitude of the compressed spectral representation are determined. Depending upon the ratio of peak-to-average magnitudes, a masked threshold is selected having a corresponding degree of tonality, and is used to determine a plurality of quantization levels and a plurality of bit allocations to perceptually encode the input audio signal with a distortion spectrum beneath a level of just noticeable distortion (JND). The invention also includes other methods and variations for selecting substantially tone-like or substantially noise-like masked thresholds for perceptual encoding of the input audio signal.

In order to implement the melody extraction or the automatic clearly more stable or to improve the transcription result, respectively, at the resulting segments or trajectories, respectively, of a melody line gained from a spectrogram of the audio signal a harmony mapping is performed such that a follower segment directly neighboring a reference segment in time direction is virtually shifted in the frequency direction by stages of octave, fifth and/or third in order to examine whether among the resulting lines of the octave, fifth and/or third there is one that fulfills a predetermined condition, like e.g. that the time/spectral representation along this line comprises a minimum which is larger by a certain factor than a minimum that it comprises along the reference segment line, and, if such a line exists, selects the same and actually performs the shifting of the follower segment. This way, errors in melody line determination may be corrected again. By a suitable selection of predetermined conditions for the lines of the octave, fifth and/or third, the direct neighborhood and further possible conditions, a faulty shifting may be prevented.

An audio encoder for encoding an audio signal, includes: a first encoding processor for encoding a first audio signal portion in a frequency domain, wherein the first encoding processor includes: a time frequency converter for converting the first audio signal portion into a frequency domain representation having spectral lines up to a maximum frequency of the first audio signal portion; a spectral encoder for encoding the frequency domain representation; a second encoding processor for encoding a second different audio signal portion in the time domain; a cross-processor for calculating, from the encoded spectral representation of the first audio signal portion, initialization data of the second encoding processor, so that the second encoding processing is initialized to encode the second audio signal portion immediately following the first audio signal portion in time in the audio signal.

An apparatus for generating a high frequency audio signal that includes an analyzer for analyzing an input signal to determine a transient information adaptively. Additionally a spectral converter is provided for converting the input signal into an input spectral representation. A spectral processor processes the input spectral representation to generate a processed spectral representation including values for higher frequencies than the input spectral representation. A time converter is configured for converting the processed spectral representation to a time representation, wherein the spectral converter or the time converter are controllable to perform a frequency domain oversampling for the first portion of the input signal having the transient information associated and to not perform the frequency domain oversampling for the second portion of the input signal not having the associated transient information.

According to an inventive scheme for introducing a watermark into an information signal, the information signal is at first transferred from a time representation to a spectral/modulation spectral representation). The information signal is then manipulated in the spectral/modulation spectral representation in dependence on the watermark to be introduced to obtain a modified spectral/modulation spectral representation, and subsequently an information signal provided with a watermark is formed based on the modified spectral/modulation spectral representation. An advantage is that, due to the fact that the watermark is embedded and/or derived in the spectral/modulation spectral representation or range, traditional correlation attacks as are used in watermark methods based on a spread-band modulation cannot succeed easily.

The time-discrete audio signal is processed (52) to provide a quantized block with quantized spectral values. Furthermore, an integer spectral representation is generated from the time-discrete audio signal using an integer transformation algorithm (56). The quantized blocks generated using the psychoacoustic model (54) are inverse quantized and rounded (58) to subsequently form differences between the integer spectral values ​​and the inverse quantized rounded spectral values. After decoding, the quantization block alone provides a lossy psychoacoustically encoded/decoded audio signal; while in decoding, the quantization block together with the combining module provides a lossless or nearly lossless encoded and decoded audio signal again. By generating a differential signal in the frequency domain, a simple encoder/decoder structure is formed.

A device and method for manipulating an audio signal includes a windower for generating a plurality of consecutive blocks of audio samples, the plurality of consecutive blocks including at least one padded block of audio samples, the padded block having padded values and audio signal values, a first converter for converting the padded block into a spectral representation having spectral values, a phase modifier for modifying phases of the spectral values to obtain a modified spectral representation and a second converter for converting the modified spectral representation into a modified time domain audio signal.

For postprocessing spectral values which are based on a first transformation algorithm for converting the audio signal into a spectral representation, first a sequence of blocks of the spectral values representing a sequence of blocks of samples of the audio signal are provided. Hereupon, a weighted addition of spectral values of the sequence of blocks of spectral values is performed in order to obtain a sequence of blocks of postprocessed spectral values, wherein the combination is performed such that for calculating a postprocessed spectral value for a frequency band and a time duration a spectral value of the sequence of blocks for the frequency band and the time duration and a spectral value for another frequency band or another time duration are used, wherein the combination is further performed such that such weighting factors are used that the postprocessed spectral values are an approximation to the spectral values as they are obtained by converting the audio signal into a spectral representation using a second transformation algorithm which is different from the first transformation algorithm. The postprocessed spectral values are in particular used for a difference formation within a scalable encoder or for an addition within a scalable decoder, respectively.

Evaluation of a battery state comprises transforming a time based history of the load on the battery into a spectral representation of that history in a load domain, e.g., the current domain. The method also comprises comparing the spectral representation to an expected battery capability for the load represented by each line in the spectra and calculating the fraction of the expected capability used at each spectral line. The method still further comprises aggregating the calculated fractions into a total fraction that represents the estimated fraction of the expected battery capability associated with that particular time history.

For embedding binary payload in a carrier signal, which, for example, is an audio signal, a sequence of time-discrete values of the carrier signal is converted to the frequency domain by means of an integer transform algorithm to obtain binary spectral representation values. Bits of the binary spectral representation values with a valency less than signal limit valency are determined and set according to the payload. The signal limit valency for a spectral representation value is less than the valency of the leading bit of this spectral representation value, so that, with adequate distance, a psychoacoustic transparent insertion of information is achieved. Thus a modified spectral representation with inserted information is generated which is finally converted back to the time domain using an integer back transform algorithm. For extracting the payload, the time-discrete signal with the inserted information is again converted to a spectral representation with the integer forward transform algorithm. Furthermore, signal limit valency information is determined to identify the bits of the binary spectral representation values containing no information regarding the carrier signal, but information regarding the payload signal, to extract these bits. The inventive concept is simple in its implementation and may be scaled with respect to the data rate of the information to be inserted.

The invention relates to a dynamic calculating method of wind-caused vibration and windage yaw of a contact net of an electrified railway. With the development of a high-speed railway, an influence degree on security of train operation by wind becomes more obvious; and effects of strong wind at a wind field and a wind gap as well as an effect of wind caused by train operation will enable large and complex vibration and windage yaw to occur at a contact net, so that a dynamic quality of high speed operation of a bow net will be influenced and bow net contact security will be threatened. The method comprises the following steps that: a wind tunnel test of a contact wire static force segment model is utilized to test an aerodynamic force parameter of a contact wire; a finite element model of wind-caused vibration and windage yaw of a contact net is established; average displacement of a contact net thread is calculated; and a spectral representation method is employed to simulate a turbulent flow wind speed time history and a time history analysis method is employed to calculate dynamic displacement of the contact net thread; and vertical average displacement and vertical dynamic displacement are superposed to obtain total displacement of wind-caused vibration of the contact net thread; and horizontal average displacement and horizontal average displacement are superposed to obtain total windage yaw of the contact net thread. According to the invention, a method is provided for dynamic calculation of wind-caused vibration and windage yaw of a contact net; and moreover, the method is suitable for application to power transmission line route.

Audio encoder for encoding a multichannel signal is shown. The audio encoder includes a downmixer for downmixing the multichannel signal to obtain a downmix signal, a linear prediction domain core encoder for encoding the downmix signal, wherein the downmix signal has a low band and a high band, wherein the linear prediction domain core encoder is configured to apply a bandwidth extension processing for parametrically encoding the high band, a filterbank for generating a spectral representation of the multichannel signal, and a joint multichannel encoder configured to process the spectral representation including the low band and the high band of the multichannel signal to generate multichannel information.

The present document relates to methods and systems for encoding an audio signal. The method comprises determining a spectral representation of the audio signal. The determining a spectral representation step may comprise determining modified discrete cosine transform, MDCT, coefficients, or a Quadrature Mirror Filter, QMF, filter bank representation of the audio signal. The method further comprises encoding the audio signal using the determined spectral representation; and classifying parts of the audio signal to be speech or non-speech based on the determined spectral representation. Finally, a loudness measure for the audio signal based on the speech parts is determined.