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WO/1993/011530A1 |
A priority assignment method (100) and device (300) are set forth for assigning a priority to a selected speech frame coded by a linear predictor coder based on at least two of: an energy of the speech frame, a log spectral distance betw...
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WO/1993/009644A1 |
The description relates to a process for reducing data in the transmission and/or storage of digital signals from several dependent channels. The dependence of the signals in a channel, e.g. in a left and right-hand stereo channel, can b...
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WO/1993/008651A1 |
A process is disclosed for reducing frequency interlacing during transmission of digitized audio signals. Frequency interlacing is always to be expected when the input signal is distributed among partial bands in series connected stages....
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WO/1993/005598A1 |
A first and second digital rate converter device and method provides for synchronization between a first digital sample sequence and a second digital output sequence related to the first digital sample sequence for a rate converter pair ...
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WO/1993/002508A1 |
In oder to find the overall monitoring threshold (6) during a bit-rate-reducing source coding of digitised audio signals (1), a requantization and coding regulation for the time or spectral scanning values (2) of the audio signal is foun...
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WO/1992/022152A1 |
A method and apparatus for acquiring and analysing PCM data in a communications system. The method comprises extracting a PCM data of a selected circuit from PCM streams (30), sampling the PCM data for one or more predetermined periods a...
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WO/1992/022060A1 |
An electronic method and apparatus for signal encoding and decoding to provide ultra low distortion reproduction of analog signals, while remaining compatible with industry standardized signal playback apparatus not incorporating the dec...
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WO/1992/016996A1 |
An audio signal has its dynamic range compressed by a system which first samples a block of the audio signal (10), typically several seconds long. The level of the signal in this block is analysed (12, 14) and an ideal signal level is ca...
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WO/1992/014339A1 |
A decoder for decoding coded video signals comprising a base layer and an enhancement layer includes a cell loss detector (66) for detecting when cells are missing from the enhancement layer. In such an event the enhancement data from th...
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WO/1992/014325A1 |
A narrow band complex input signal, typically having multiple independently modulated carriers, is processed to produce an output signal. The process separates the multiple input signals (10) into an amplitude-limited angle-modulated car...
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WO/1992/013396A1 |
The transmission of digitised, block-coded audio signals makes use of scale factors formed in the block-coding of the digitised audio signals from the peak value of a sequence (block) of signal values and added to the scanned and possibl...
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WO/1992/012608A1 |
The invention relates in general to the reproducing of multi-channel signals. More particularly, the invention relates to the decoding of multi-channel audio signals representing multidimensional sound fields delivered by one or more del...
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WO/1992/012607A1 |
The invention relates in general to the recording, transmitting, and reproducing of multidimensional sound fields intended for human hearing. More particularly, the invention relates to subband encoding and decoding of signals representi...
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WO/1992/010037A1 |
A decoder for recovering an analog signal from its digital representation produced by an encoder employing adaptive-delta-modulation first decodes audio bit-stream (17) in the digital domain with digital delta demodulation (15), to a pul...
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WO/1992/008289A1 |
DE 37 02 490 describes a process for data transmission in which data from pixel blocks are converted into digital signals which are coded with variable word length. Here the digital signals are displayed in blocks of an average length. I...
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WO/1992/006467A1 |
An electronic system and a method substantially eliminates any delays caused by buffering of a signal containing speech. The electronic system comprises a receiver for receiving signals (16), a buffer (26), coupled to the receiver, a buf...
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WO/1992/005541A1 |
A voice coding system which finds by evaluation operation a code vector that minimizes the error between an input voice signal and a reproduced signal obtained through a linear estimation synthesis filtering simulating the vocal tract ch...
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WO/1992/000637A1 |
For individual tuning of the dynamics of a radio programme signal (variable dynamics), a control parameter is derived from the associated radio programme signal and emitted together with the programme signal. At the receiver, the control...
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WO/1991/018361A1 |
The disclosure involves the use of a library of modulated wavelet-packets which are effective in providing both precise frequency localization and space localization. An aspect of the disclosure involves feature extraction by determinati...
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WO/1991/018464A1 |
A pi/4-shift DQPSK modulator modulates a digitized voice signal and other information. An FM modulator modulates the analog voice signal and other information. The FM modulator is coupled to the quadrature mixers (109 and 110) of the pi/...
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WO/1991/016769A1 |
This invention relates in general to high-quality low bit-rate digital transform coding and decoding of information corresponding to audio signals such as music signals. More particularly, the invention relates to signal analysis/synthes...
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WO/1991/015926A1 |
In a method of aligning a data receiver (2) with a data transmission from, e.g., a head-end station (1), data is encoded for transmission with a hybrid differential coding scheme including at least one invalid codeword, and the coding sc...
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WO/1991/013370A1 |
The figure shows a transmitter (10), where an oscillator (12) generates a base frequency of 100 kHz to be fed into a divide-by-four unit (14). The divide-by-four unit (14) produces a 25 kHz output signal to be fed into the regulated powe...
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WO/1991/010283A1 |
The invention relates to an interpolative analog/digital converter for band-pass signals in which the amplitudes of the scanning values are quantized in 2 consecutive stages. The first stage contains the actual analog/digital interface a...
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WO/1991/005412A1 |
The process disclosed enables the ''time domain aliasing cancellation'' method to be extended systematically to larger block overlapping. The boundary conditions which, when using various modified transforms, the analysis and synthesis w...
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WO/1991/005411A1 |
A signal, preferably an audio signal, is subdivided by windows into successive blocks each of which is converted into a spectrum by transform. The spectra resulting from the transform are then coded, transmitted, decoded after transmissi...
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WO/1991/003901A1 |
A relay exchange system in which a plurality of switchboards for switching transmission information contained in frames each consisting of m bits are connected in cascade using a plurality of transmission paths, a pair of multiplexers ar...
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WO/1990/016128A1 |
A signal transmission and/or storage system comprising a transmission stage and, after transmission and/or storage of signals transmitted on a limited band-pass medium involving processing to reduce the volume of information to be transm...
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WO/1990/014719A1 |
A signal, in particular an audio signal, is divided by means of windows into consecutive blocks which overlap by at least 50 % and the signal sections contained in the blocks are evaluated by means of analysis windows. The partial signal...
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WO/1990/011650A1 |
Electrical circuits suitable for encoding binary information, in accordance with a novel modulation method. The novel modulation method is referenced in the instant case, and it is explained that the method may be used when the encoding ...
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WO/1990/010980A1 |
A time domain communications system (10, 96) wherein time modulated (220), impulse derived signals are multiplied (226) by a template signal (234), integrated (248), and then demodulated (254, 258, 260). By this process, usable signals (...
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WO/1990/010993A1 |
A system for transmitting and receiving multiplexed video and audio signals. The video and audio signals are balanced to enhance the transmission quality and to use the transmission capability fully under the condition of a limited trans...
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WO/1990/009063A1 |
In a process for transmitting a signal which is subdivided by windows into consecutive, overlapping blocks, each partial signal contained in the blocks is converted by transformation into a spectrum, the sepctra are then coded, transmitt...
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WO/1990/009022A1 |
A transform encoder/decoder system employs a discrete transform, carefully designed analysis/synthesis window pairs, adaptive transform coefficient quantization, and adaptive bit allocation to encode high-quality music including broadcas...
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WO/1990/009064A1 |
A low bit-rate (192 kBits per second) transform encoder/decoder system (44.1 kHz or 48 kHz sampling rate) for high-quality music applications employs short time-domain sample blocks (128 samples/block) so that the system signal propagati...
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WO/1990/006632A1 |
A method of processing data for transmission from a transmitter (T) to a receiver (R) both of which are switchable between a compression mode in which the data stream is encoded or decoded with a data compression algorithm and a transpar...
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WO/1990/004890A1 |
A circuit for demodulating digital voice signals to prevent generation of large voice distortion when the interpolation is effected continuously to correct error in case error has developed continuously. The demodulation circuit comprise...
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WO/1990/004891A1 |
A circuit for demodulating digital voice signals to suppress the broken noise that generates when the synchronism is no more maintained or when the mute is forcibly applied. The demodulation circuit comprises a synchronism detecting circ...
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WO/1990/004873A1 |
A process for varying the load on a power supply system by injecting signals into the system using an audio frequency signal generator in which the time interval between the signals is reduced to reduce the load on the system and increas...
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WO/1989/011759A1 |
The audio signal to be transmitted is divided into chronologically consecutive blocks through time slots, then the signal portions included in the blocks are converted into a short-time spectrum, thereafter the short-time spectra are cod...
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WO/1989/011761A1 |
Apparatus and method for analyzing a bit stream to determine whether the stream is encoded in A-law PCM or mu-law PCM. Bit positions (2, 3, and 4) are examined to determine which of the 8 possible segment codes are present. Segment codes...
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WO/1989/010661A1 |
According to the invention, an information signal is sampled, quantized and processed digitally through an information encoder (100). Digitally processed samples (1104, 1106) produced by the encoder (100) are modulated onto a communicati...
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WO/1989/010031A1 |
A method and apparatus for transmitting information comprises a signal generator (1) for continuously generating a pair of optical signals (f1, f2). These signals are similarly modulated in a modulator (5) with information to be transmit...
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WO/1989/008357A1 |
The device described has the following features: the device has a first input terminal to which is applied the input signal of the acoustic signal processing system to be monitored, a second input terminal to which is applied the output ...
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WO/1989/007866A1 |
In a method and apparatus for coding high quality digital audio signals, the signal is sampled at repeated intervals and the sample is filtered into a number of sub-bands, preferably four. The signal in each sub-band is coded by means of...
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WO/1989/007368A1 |
An improved apparatus (112) and method for sigma delta converter for bandpass signals is disclosed, suitable for use in mobile radio applications, between the front end and digital signal processing states, that includes at least one ban...
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WO/1989/007372A1 |
Variable length codewords are grouped into successive N bit word groups (25) in sequential order as received from a variable length codeword encoder (18). The N-bit groups are each stored in corresponding N-bit-wide memory locations in a...
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WO/1989/006418A1 |
Input speech is processed to derive (2) LPC filter parameters and (5) parameters of a multipulse excitation which are quantised (10) prior to transmission along with the filter parameters to a decoder where the excitation is generated an...
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WO/1989/003574A1 |
In a digital adaptive transform coding process for transmitting and/or storing acoustic signals, in particular musical signals, scanning values N of the acoustic signals are transformed into spectral coefficients M which are divided into...
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WO/1989/003140A1 |
During transformative coding of audio signals, the bits are allocated to the spectrum components for purposes of quantification. After a preliminary approximative quantification, there remains the problem of distributing the still availa...
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