摘要:
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.
摘要:
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.
摘要:
MP3 decoders decode MP3 data streams that comprise headers and signal data interspersed with each other, each header specifying a distance to a subsequent header, each header corresponding to a frame of signal data, the header being associated with a pointer that points to a starting point of the signal data for that frame relative to the header. An editing system cuts tracks from existing data streams. During editing, a user signals a header corresponding to the start of the desired track. The track is from the data stream, including a part of the data stream pointed at by the header and preceding the specified header. A new MP3 compatible data stream is written to a medium. The new data stream contains said header as first valid header and said part of the data stream preceding the header.
摘要:
The present invention relates to a method, device (12) and computer program product for enabling detection of additional data embedded in a media signal that may have been subjected to scaling. The invention also relates to an additional data detecting device (10) comprising such a device for enabling detection. An envelope discriminating unit (ED) provides a first extracted narrow band envelope signal sample (we[n]) from an input media signal sample (yb[n]), and a variable scale down sampling unit (VSDS) down samples the narrow band envelope signal sample using a down sampling rate that is dependent on a scaling factor variable value (η) for providing at least one sample of a first additional data estimate (wn[k]) in order to allow the detection of additional data in said signal sample.
摘要:
A method of encoding input signals (1, r) to generate encoded data (100) is provided. The method involves processing the input signals (1, r) to determine first parameters (Õ 1 , Õ 2 ) describing relative phase difference and temporal difference between the signals (1, r), and applying these first parameters (Õ 1 , Õ 2 ) to process the input signals to generate intermediate signals. The method involves processing the intermediate signals to determine second parameters (±; IID, Á) 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. The method also involves quantizing the first parameters, the second parameters, and dominant and residual signals (m, s) to generate corresponding quantized data for subsequent multiplexing to generate the encoded data (100).
摘要:
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.
摘要:
A transmitter is disclosed for transmitting a digital information signal having a specific first sampling frequency (f s1 ) via a transmission medium. The digital information signal is lowpass filtered (10) and down sampled (12) so as to obtain a low frequency component of the digital information signal. This signal is transmitted via the transmission medium (TRM). Further the low frequency component is upsampled (22) and filtered (28) and subsequently subtracted (34) from the original digital information signal. The difference signal thus obtained is also transmitted.
摘要:
A method using synthetic noise sources in a multi-channel audio coding system for encoding a set of audio signals 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 the 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.
摘要:
An audio signal is encoded by a first waveform encoder (103) to generate a first waveform based bit-stream component. A second encoder (105) encodes the audio signal to generate a second bit-stream component comprising first enhancement data and a third encoder (107) encodes the audio signal to generate a third bit-stream component comprising second enhancement data for the first waveform based bit-stream component. The first and second bit-stream components correspond to a first representation of the audio signal and the first and third bit-stream components correspond to a second representation of the audio signal. A scalable audio bit-stream is generated by a bit-stream generator (109). The different representations may be selected between by a decoder thereby allowing a flexible and scalable bit-stream to be communicated. The second encoder (105) may specifically be a waveform encoder and the third encoder (107) may specifically be a parametric encoder.
摘要:
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, 12.0) 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 resealed signal (Y').