Abstract: Human speech is coded by singling out from a transfer function of the speech, all poles that are unrelated to any particular resonance of a human vocal tract model. All other poles are maintained. A glottal pulse related sequence is defined representing the singled out poles through an explicitation of the derivative of the glottal air flow. Speech is outputted by a filter based on combining the glottal pulse related sequence and a representation of a formant filter with a complex transfer function expressing all other poles. The glottal pulse sequence is modelled through further explicitly expressible generation parameters. In particular, a non-zero decaying return phase supplemented to the glottal-pulse response that is explicitized in all its parameters, while amending the overall response in accordance with volumetric continuity.

Abstract: For coding human speech for subsequent audio reproduction thereof, a plurality of speech segments is derived from speech received, and systematically stored in a data base for later concatenated readout. After the deriving, respective speech segments are fragmented into temporally consecutive source frames, similar source frames as governed by a predetermined similarity measure thereamongst that is based on an underlying parameter set are joined, and joined source frames are collectively mapped onto a single storage frame. Respective segments are stored as containing sequenced referrals to storage frames for therefrom reconstituting the segment in question.

Abstract: A method is described for short-time Fourier-converting a speech signal and for resynthesizing an output speech signal from the modulus of its short-time Fourier transform and from an initial phase. In particular, after the Fourier converting the signal is subjected to a phase-specifying operation. Subsequently speech duration is affected by systematically maintaining, periodically repeating or periodically suppressing result intervals of the successive Fourier converting and phase affecting. Finally, a resynthesizing operation is executed. Speech pitch can likewise be affected through systematically excising or inserting signal intervals. Finally, the two strategies can be combined, so that ultimately, pitch and duration can be affected independently from each other.

Abstract: For encoding sound received as a stream of multibit input sample, from a finite length sequence of input samples, an instantaneous audibility threshold characteristic is determined. Next, a shaped, dither determined, signal is subtracted from the input samples to produce processing samples. Subtracting a dither signal dynamically ignores processing sample bits below the threshold. Next, quantization by a variable number of bits below the threshold is performed, while retaining all sample bits above the threshold. The ignored bits are replaced by the dither signal as buried channel bits at an adjustable number of bits per sample. Therefore, upgraded samples have non-ignored bits and buried channel bits are obtained and outputted. The noise is shaped through shape-filtering a difference between the upgraded samples and the processing samples, which shape-filtering may amend a received white-noise-like signal towards an actual threshold-versus-frequency characteristic.

Abstract: Subband coding a digital signal having a first and a second signal component in a stereo intensity mode. The digital signal is subband coded to produce a first subband signal having a first q-sample signal block in response to the first signal component, and a second subband signal having a second q-sample signal block in response to the second signal component, the first and the second subband signals being in the same subband, and the first and second signal blocks being time-equivalent. The first and second signal blocks are processed to obtain a minimum distance value representative of a distance between a line and a plurality of points if (a) the points correspond to the respective pairs of time-equivalent samples in the first and second signal blocks, and are plotted in a coordinate system, and (b) the line is plotted such that it traverses the origin and the points at a minimum distance from the points in the coordinate system.

Abstract: For subband coding of a signal such as a digital stereo signal, the digital signal is split into subbands each comprising at least first and second subband signal components. The subband signal components in a subband m are constituted by blocks of q samples each, represented by a certain number of bits (n.sub.mr, n.sub.m1). The number of bits is the result of allocation based on a determination of bit needs for blocks in each subband. A determination is made whether or not initially bits are to be allocated to a signal block of one component in a subband prior to the bit allocation, and irrespective of the bit need for that signal block. If it is determined that an initial bit allocation is required for a particular signal block in a subband, an initial bit allocation is also applied to at least a second corresponding signal block in that subband.

Abstract: Transmitter, encoding system and method for subband coding a digital signal. The encoding system includes a splitter unit for dividing the digital signal into subband signals SB.sub.1, . . . , SB.sub.p ; a quantizer unit for quantizing time-equivalent q sample signal blocks of the subband signals; a bit need determining unit and a bit allocation unit. The bit need determining unit determines a bit need b.sub.m which corresponds to the number of bits by which the q samples in a time-equivalent signal block in a subband signal SB.sub.m should be represented, where 1.ltoreq.m.ltoreq.P. The bit allocation unit allocates n.sub.m bits to each of the q samples of the time-equivalent signal block of subband signal SB.sub.m on the basis of the bit need b.sub.m and an available bit quantity B, n.sub.m being the number of bits by which the q samples in the time-equivalent signal block of subband signal SB.sub.m will be represented, where 1.ltoreq.m.ltoreq.P.

Abstract: Transmitter, encoding system and method for subband coding a digital signal. The encoding system includes a splitter for dividing the digital signal into subband signals SB.sub.1, . . . , SB.sub.M ; a quantizer unit for quantizing time-equivalent q sample signal blocks of the subband signals; a bit need determiner and a bit allocator. The bit need determiner determines a bit need b.sub.m which corresponds to the number of bits by which the q samples in a time-equivalent signal block in a subband signal SB.sub.m should be represented, where 1.ltoreq.m.ltoreq.M. The bit allocator allocates n.sub.m bits to each of the q samples of the time-equivalent signal block of subband signal SB.sub.m on the basis of the bit need b.sub.m and an available bit quantity B, n.sub.m being the number of bits by which the q samples in the time-equivalent signal block of subband signal SB.sub.m will actually be represented, where 1.ltoreq.m.ltoreq.M.

Abstract: A method and apparatus for replacing invalid samples of an equidistantly sampled signal with corrected samples derived from an environment of valid samples preceding and following the invalid samples. The values of the corrected samples are estimated by first determining a quantity q representing a number of samples corresponding to the periodicity of a segment of the signal over an interval including valid samples and the invalid samples. Such quantity q is utilized in a system of autoregression equations in which the corrected sample values are unknown quantities, and such equations are solved by a microprocessor to derive estimates of such sample values on a "real time" basis. The method can be applied to achieve restoration of a speech signal in which there may be intervals of up to approximately 12 msec containing invalid samples.

Abstract: A digital transmission system is disclosed having a transmitter (3,6,9) and a receiver (13,16,19) for transmitting a digital signal, such as a digital audio signal, having a given sampling rate F.sub.S. The digital signal is subband coded into M subbands with sampling rate reduction. To that purpose the transmitter includes a first unit (3) for splitting up the digital signal into M signals having a sampling rate F.sub.S /M. These signals are available at the outputs (4.1 to 4.M) of the first unit (3), and are applied to M analysis filters (6.1 to 6.M), each filter (6.m) having one input (5.m) and two outputs (7.ma and 7.mb). The 2M filter outputs are coupled to 2M inputs (8.1 to 8.2M) of a signal processing unit 9 which has M outputs (10.1 to 10M) for supplying the M subband signals (S.sub.1 to S.sub.M). Each filter (6.m) is adapted to apply two different filterings on the signal applied to its input and to supply the two different filtered versions of the input signal to its two outputs.

Abstract: The audio signal is divided into frequency sub-bands, the sub-band samples are quantized according to a predetermined quantizing criterion, and the quantized samples in one or more sub-bands are summed with samples of the auxiliary signal in such sub-bands, the auxiliary signal sample in a sub-band having a maximum amplitude less than half the quantization step used in such sub-band. The combined signal, covering the entire frequency band of the audio signal, can be transmitted or recorded on a record carrier. Upon audio reproduction at a receiver only the audio signal will be audible, the auxiliary signal being masked due to the psychoacoustic characteristics of the human auditory system which are unresponsive to low-level noise in the presence of high amplitude sound. The receiver includes a decoder which analyzes the combined signal into the original frequency sub-bands and re-quantizes the sub-band signals using the same quantizing criterion as at the coder.

Abstract: A system for transmitting digitized analog information includes, for the purpose of transmitting an additional information signal apparatus for modifying the bit of low significance of every n-th data word representing the digitized information depending on the bits of an additional information signal. The apparatus for modifying is constructed in such a way that the modification depends both on the additional information signal and on other bits such as the most significant bit b7 of the data word which are not correlated with the bits of the additional information signal. This results in the intercorrelation between the errors in the data words introduced by the modification being reduced, so that a flatter power density error spectrum is obtained and the errors introduced in the data words by the modification are less conspicuous. A system for recovering the additional information signal from received modified data words includes similar apparatus for reversing the modification.

Abstract: A system for subband coding of a digital audio signal x(k) includes in the coder (1) a filter bank (3) for splitting the audio signal band, with sampling rate reduction, into subbands (p=1, . . . P) of approximately critical bandwidth and in the decoder (2) a filter bank (5) for merging these subbands, with sampling rate increase. For each subband (p) the coder (1) comprises a detector (7(p)) for determining a parameter G(p;m) representative of the signal level in a block (p;m) of M samples of the subband signal x.sub.p (k) as well as a quantizer (8(p)) for adaptively block quantizing this subband signal in response to parameter G(p;m), and the decoder (2) comprises a dequantizer (9(p)) for adaptively block dequantizing the quantized subband signal s.sub.p (k) in response to parameter G(p;m).

Abstract: The values of incorrect samples of, for example, a digital audio signal are estimated by means of interpolation. For this purpose an appropriate sampling interval is determined from the number of incorrect samples, in which interval the incorrect samples are situated. Subsequently, a best-fitting recursion formula is determined from the values of the samples, a first estimate being taken for the values of the incorrect samples. By means of this recursion formula the value of a sample is expressed as a weighted sum of the values of a number of preceding samples and an error term. By means of the recursion formula thus found the values of the incorrect samples are subsequently estimated, in such a manner that they are best-fitting values for the recursion formula. The Application also discloses a device for carrying out the method.

Abstract: A device for reproducing digitally presented video pictures includes a restoration circuit for restoring erroneous picture elements. To this end a reference sub-picture is composed of erroneous and correct picture elements, the erroneous picture elements being entirely enclosed by the correct picture elements. Before restoring the erroneous picture elements, a correction function is determined, starting from the correct picture elements of the reference sub-picture, and by means of this correction function a correction value can be determined for each correct picture element in the reference sub-picture such that the difference picture elements obtained by subtracting the associated correction value from each correct picture element represent a signal whose mean value is zero. A known method of restoring erroneous picture elements can now be used for these difference picture elements.

Abstract: A system for subband coding of a digital audio signal x(k) includes in the coder (1) a filter bank (3) for splitting the audio signal band, with sampling rate reduction, into subbands (p=1, . . . P) of approximately critical bandwidth and in the decoder (2) a filter bank (5) for merging these subbands, with sampling rate increase. For each subband (p) the coder (1) comprises a detector (7(p)) for determining a parameter G(p;m) representative of the signal level in a block (p;m) of M samples of the subband signal x.sub.p (k) as well as a quantizer (8(p)) for adaptively block quantizing this subband signal in response to parameter G(p;m), and the decoder (2) comprises a dequantizer (9(p)) for adaptively block dequantizing the quantized subband signal s.sub.p (k) in response to parameter G(p;m).

Abstract: A device for reproducing digitally presented video pictures includes a restoration circuit for restoring erroneous picture elements. To this end a reference sub-picture is composed from the erroneous picture elements and non-erroneous picture elements, the erroneous picture elements being completely enclosed by the non-erroneous picture elements. The erroneous picture elements are subsequently replaced by zero picture elements. Each picture element is then considered as a sum of weighted versions of picture elements which form part of a prediction field. The set of prediction coefficients associated with this prediction field is obtained by minimizing a prediction coefficient function. This operation is performed for different prediction fields and the restoration process is continued in known manner, this continuation being based on the prediction field yielding the smallest value for the prediction coefficient function.

Abstract: Method and apparatus for replacing a sequence of invalid samples of an equidistantly sampled signal, such as an audio signal, with valid signal samples from a remote preceding environment of valid samples which, but for a scaling factor, approximates as closely as possible the environment preceding the set of invalid samples to be replaced. Although a sequence of valid samples subsequent to the set of invalid samples is necessary to define the environment of the invalid set, the subsequent sequence can be relatively short. Consequently, the time delay required for replacing invalid samples is much shorter than with methods based on valid samples in an environment subsequent to the invalid samples.