Abstract: A device (1) for adding reverberation to an input signal (s(n)) uses a corresponding transformed input signal (S(n)), for example the Fourier transform of the signal, to generate reverberation. The device comprises circuits (3, 5, 6) for combining the transformed input signal (S(n)) with a modified and delayed transformed output signal (G(Sr(n?i))) so as to produce a current transformed output signal (Sr(n)). Inversely transforming the current transformed output signal (Sr(n)) results in an output signal (sr(n)) having reverberation.

Abstract: There is provided a watermark detector (20) including an input for receiving an input signal (Y?) including watermark content (W) to be searched. A first processor (40) of the detector (20) is operable to analyse portions (100, 110, 120) of the signal (Y?) to identify corresponding sets of characteristic properties or fingerprints (P1 to Pq) and associated temporal descriptors (d1 to dq). A communication link to a database (50) is provided for communicating the fingerprints to the database (50) to identify the signal and to determine corresponding temporal descriptors (MT1 to MTq) corresponding to the portions (100, 110, 120) in the original signal. A second processor (220) is included for calculating from a difference between the temporal descriptors (d1 to dq) and the retrieved temporal descriptors (MT1 to MTq) a scaling factor to which the input signal (Y?) has been subjected. The scaling factor is useable for re-scaling the signal and extracting the watermark from the rescaled signal (Y?).

Abstract: A method of encoding input signals (l, r) to generate encoded data (100) is provided. The method involves processing the input signals (l, r) to determine first parameters (?1,?2) describing relative phase difference and temporal difference between the signals (l, r), and applying these first parameters (?1, ?2) to process the input signals to generate intermediate signals. The method involves processing the intermediate signals to determine second parameters (?; IID,?) describing angular rotation of the first intermediate signals to generate a dominant signal (m) and a residual signal (s), the dominant signal (m) having a magnitude or energy greater than that of the residual signal (s). These second parameters are applicable to process the intermediate signals to generate the dominant (m) and residual (s) signals.

Abstract: A stereo audio encoder (100) comprises a parametric stereo encoder (115) which generates a mono signal and parametric stereo parameters for at least a high frequency part of an input stereo signal. A stereo intensity encoder (117) generates stereo intensity data for the mono signal. The mono signal and intensity data are encoded in accordance with an encoding standard such as MPEG Layer II and the parametric stereo parameters are included in the ancillary data sections by an output processor (113). Thus, a legacy decoder (such as an MPEG Layer II decoder) may generate a stereo signal using the stereo intensity data whereas a higher complexity decoder may generate a high quality audio signal using the parametric stereo parameters. A stereo decoder (200) receives the encoded data from the encoder (100). An intensity decoder (203) generates a stereo signal using intensity data. This is fed to a parametric stereo decoder (207) which processes the stereo signal in accordance with extracted parametric stereo data.

Abstract: The invention relates to a record carrier carrying a stereo signal and a data signal. The data signal comprises a first portion and a remaining portion. A first channel on the record carrier comprises a composite signal. The composite signal is obtained by combining the stereo signal and the first portion of the data signal, preferably by using buried data techniques. A second channel on the record carrier comprises the remaining portion of the data signal. The invention further relates to a method, a recording apparatus and a reproducing apparatus for recording/reproducing the stereo signal and the data signal on a record carrier.

Abstract: Synthesizing an output audio signal is provided on the basis of an input audio signal, the input audio signal comprising a plurality of input sub-band signals, wherein at least one input sub-band signal is transformed (T) from the sub-band domain to the frequency domain to obtain at least one respective transformed signal, wherein the at least one input sub-band signal is delayed and transformed (D, T) to obtain at least one respective transformed delayed signal, wherein at least two processed signals are derived from the at least one transformed signal and the at least one transformed delayed signal, wherein the processed signals are inverse transformed (T?1) from the frequency domain to the sub-band domain to obtain respective processed sub-band signals, and wherein the output audio signal is synthesized from the processed sub-band signals.

Abstract: The invention relates to a signal encoding system (100). A pre-encoder (103) encodes a signal and generates a pre-encoded signal. In addition the pre-encoder (103) generates encoding assistance data which is stored in a signal storage (105) together with the pre-encoded signal. When the signal is retrieved from the signal storage (105), it is decoded in a decoder (111) and a watermark is inserted in the decoded signal to generate a watermarked signal. The watermarked signal is then re-encoded, possibly at a different encoding rate, in a re-encoder (117). The re-encoder (117) is operable to re-encode the watermarked signal in response to the encoding assistance data. Thus, encoding assistance data may be generated at encoding prior to storage and the encoding assistance data may be used to facilitate re-encoding of a watermarked signal when retrieved.

Abstract: An output audio signal (L, R) is generated based on an input audio signal, the input audio signal comprising a plurality of input subband signals (N). The input subband signals are delayed in a plurality of delay units (76) to obtain a plurality of delayed subband signals, wherein at least one input subband signal is delayed more than a further input subband signal of higher frequency, and wherein the output audio signal is derived (77) from a combination of the input audio signal and the plurality of delayed subband signals.

Abstract: There is provided a method of allocating optimal payload space in a watermarking system. The method includes steps of: (a) obtaining identification parameters relating to programme data content (PC), said identification parameters including at least one or more user identifiers (UID) and one or 5 more programme content identifiers (CID); (b) storing said identification parameters (UID, CID) in one or more databases (30); (c) generating one or more transaction numbers (nTR) capable of being uniquely mapped to corresponding identification parameters (UID, CID) stored in the one or more 10 databases (30); and (d) generating watermark information for carrying said one or more transaction numbers (nTR) and embedding said watermark information as an optimized payload (OPL) into said programme data content (PC) to generate corresponding watermarked data content (WPC).

Abstract: The present invention relates to methods, devices, a signal and recorded medium relating to a fragile watermark (WM) in a media signal. According to the invention a fragile watermark is provided in or in coding related to a buried data channel (40, 42, 44, 46) provided in the audio samples of the media signal, In this way the fragile watermark can be made to be very complex and still remain imperceptible to a user.

Abstract: The invention relates to a method of enabling a user to adjust at least a first and a second control parameter for controlling an electronic system. The method comprises the step of displaying a coordinate system on a display screen, a first coordinate of said coordinate system representing a range of values of the first control parameter, and a second coordinate of said coordinate system representing a range of values of the second control parameter, the step of visually indicating a position in said coordinate system corresponding to a currently selected combination of values of the first and the second control parameter, and the step of enabling the user to select a new combination of values of the first and the second control parameter by indicating a position within said coordinate system.

Abstract: Method and apparatus are described for compensating for a linear time scale change in a received signal, so as to correctly rescale the frame sequence of the received signal. Firstly, an initial estimate of the sequence of symbols is extracted from the received signal. Successive estimates of correctly time scaled sequences of the symbols are then generated by interpolating the values of the initial estimates.

Abstract: Method and apparatus are described for compensating for time offset in a received signal, so as to correctly align the frame sequence of a received signal to a sequence of transmitted symbols. Each symbol extends over Ts signal samples. The received signal is first divided into a sequence of frames of length Ts, and then each framedivided into a multiplicity of Nb sub-frames. Subsequently, Nb sequences of values are formed, where every successive value in each sequence is derived from the corresponding sub-frame within each successive frame. Each of the Nb sequences is an estimate for the correctly aligned sequence of transmitted symbols.

Abstract: Sum/difference coding of a compatible signal, typically in case of a dominant centre signal or dominant surround situation of a multi-channel audio stream to be decoded by both a stereo decoder and by a multi-channel decoder, to provide improved encoding of multiple input signals employing compatibility matrixing.

Abstract: An audio encoding scheme or a stream that encodes audio and video data is disclosed. The scheme has particular application in mezzanine-level coding in digital television broadcasting. The scheme has a mean effective audio frame length F that equals the video frame length 1/fV over an integral number M video frames, by provision of audio frames variable in length F in a defined sequence where length=F(j) at encoding. The length of the audio frames may be varied by altering the length of overlap between adjacent frames in accordance with an algorithm that repeats after a sequence of M frames. An encoder and a decoder for such a scheme are also disclosed.

Abstract: A method using sources in a multi-channel audio coding system for encoding a set of audio signals wherein correlated components are present. The method comprises the step of determining, from the relation between said audio signals, a composition of said sources, the composition being such that the sources in said composition are substantially uncorrelated, so that said composition of sources synthesizes said components in a relation-preserved way. The method may further comprise the step of encoding the sources, by determining for each source a set of parameters for synthesizing said source and a set of transformation parameters for generating said composition of sources.

Abstract: A method using synthetic noise sources in a multi-channel audio coding system for encoding a set of audio signal wherein correlated noise components are present. The method comprises the step of determining, from the relation between said audio signals, a composition of noise sources, the composition being such that the noise sources in said composition are mutually uncorrelated, so that said composition of noise sources synthesizes said noise components in a relation-preserved way. The method may further comprise the step of encoding die noise sources, by determining for each noise source a set of noise parameters for synthesizing said source and a set of transformation parameters for generating said composition of noise sources.