Sony United Kingdom Limited

Sony Corporation

Other

Yamaha Corporation

Oxford Digital Limited

Sony Europe Limited

Peter Charles Eastty

Peter Damien Thorpe

Christopher Sleight

William Edmund Kentish

Conrad Charles Cooke

Rodney Hugh Densham

Christopher Michael McCulloch

William Edmund Cranstoun Kentish

James Andrew Scott Angus

Paul Anthony Frindle

Tetsuya Konishi

Nathan Bentall

Masahiro Shimizu

Ken Iwayama

Nicholas George Tembe

Eamon Hughes

Nicholas John Haynes

Michael John Hunter Smith

A method of generating a desired shelving filter for audio data, the desired shelving filter having a low-corner frequency U and a high-corner frequency f, wherein the difference between the gain of the frequency response of the desired shelving filter at fand the gain of the frequency response of the desired shelving filter at U is substantially equal to a non-zero desired amount, the method comprising: determining an order for a first shelving filter of a first predetermined type, wherein the low-corner frequency of the first shelving filter is U and the high-corner frequency of the first shelving filter is f&#x3a3;, determining, for a second shelving filter of a second predetermined type and of a predetermined order m, a frequency fz for the low-corner frequency of the second shelving filter and a frequency Zfor the high-corner frequency of the second shelving filter, where h is at least fi, fis greater than f, and fis at most fi, and forming the desired shelving filter as a filter combination of the first shelving filter and the second shelving filter; wherein fz, fand the order of the first shelving filter are determined such that difference between the gain of the frequency response of the filter combination at fe and the gain of the frequency response of the filter combination at fi is substantially equal to the desired amount.

Audio processing

A method of processing an input audio signal, the method comprising forming a plurality of output audio signals from the input audio signal, wherein each output audio signal is formed by performing respective processing on the input audio signal, wherein for a first output audio signal there is a target audio equalization operation comprising a target filter twice, wherein for the first output audio signal, the respective processing comprises a first audio equalization operation, the first audio equalization operation being the target audio equalization operation modified to compensate for phase shifts that correspond to zeros of the transfer function of the target audio equalization operation, wherein for each output audio signal other than the first output audio signal, the respective processing comprises a compensation filter that compensates for phase shifts that correspond to poles of the transfer function of the target audio equalization operation.

Equalization of an audio signal

A computer-implemented method of determining a configuration for an audio processing operation, wherein the audio processing operation comprises a predetermined set of one or more audio processing sub-operations, each audio processing sub-operation being configurable with one or more respective configuration parameters, the method comprising: specifying the predetermined set of one or more audio processing sub-operations; specifying a target frequency response; and performing a convergent optimization process to determine a configuration for the audio processing operation that reduces a difference between the frequency response of the audio processing operation and the target frequency response, wherein the configuration comprises a respective value for each configuration parameter of each audio processing sub-operation.

Determining a configuration for an audio processing operation

In an audio network system constructed from a main node and a plurality of satellite nodes each having a plurality of ports, the main node generates and transmits a main packet including audio signals of a plurality of channels. Each satellite node selects one of the plurality of ports in turn, and confirms whether the main packet arrives at the selected port every predetermined period or not. When the main packet arrives at the selected port every predetermined period, the main packet is received via the port by continuing selection of the port. In the case where reception of the main packet is lost, one is selected from the plurality of ports in turn, the operation to confirm arrival of the main packet is restarted, and another port at which the main packet arrives at the present stage is automatically found.

Audio network system and method of detecting topology in audio signal transmitting system

An original digital audio signal represents unimpaired audio information. The digital audio signal is compressed and encrypted without substantial impairment to produce a first audio signal. A second, unencrypted, audio signal is produced. The first and second audio signals are combined. Preferably the first audio signal is losslessy compressed. The second audio signal may be an impaired version of the original digital audio signal.

Processing digital data

An analog-to-digital conversion arrangement converting an input analog signal into an output digital representation. Two or more analog-to-digital conversion paths each applying a conversion mapping between input analog signal magnitudes and respective digital values generate an intermediate representation of the input analog signal, the conversion paths being operable to apply different respective conversion mappings. An output circuit combines the intermediate representations from at least two conversion paths to generate the output digital representation, the intermediate representations being combined according to a weighting dependent on the magnitude of the input analog signal. At least one of the conversion paths has an enhanced sensitivity mode appropriate to a range of magnitudes of the input signal that are below a threshold magnitude. Control logic inhibits operation in the enhanced sensitivity mode if the magnitude of the input analog signal exceeds the threshold magnitude.

Analogue to digital conversion

Data processing apparatus comprises an array of user-operable controls, the controls being adjustable by movement of a user's hand while touching a control; a detector for detecting when a user's hand is touching a control; a display screen for displaying respective screen icons associated with the controls; and a display processor, responsive to a detection that a user's hand is touching one of the controls, for altering the screen icon associated with that control.

Data processing

Process for accurately timed automatic alteration of electrical signals stored on a signal carrier () during the transmission of the original signal to a signal receiver () in which process, for the exact dynamic alteration of the original signals, the original signals are first digitized and, provided with a time code, supplied to a computer () and stored in its memory unit, then the original signals digitally stored in the computer () are processed independently of the original recording in that the original signals digitally stored in the computer are graphically represented and checked for spots which are to be altered, and the correction signals corresponding to the desired alterations are selected and, provided with a time code, are entered in the computer () and then these correction signals are synchronously supplied to the setting members () during the transmission of the original signals from the signal carrier () to the signal receiver ().

Automation of signal processing apparatus

Signal processing apparatus is described in which at least two sets of automated signal processing functions are controlled by stored automation commands, at times dependent upon at least two respective timecode signals which may each be associated with a respective source signal supplied to the signal processing apparatus.

An audio signal processor which modifies audio signal components outside the conventional audio frequency band. The processor includes a Delta Sigma Modulator (DSM) that receives a non-interpolated digital audio signal sampled at a frequency of at least 198 kHz.

Audio signal processors

Apparatus for storing or transmitting a one-bit digital signal comprises an input inverter for inverting a subset of the data bits of an input one-bit digital signal, to generate a bit-inverted signal; a storage or transmission medium for storing or transmitting the bit-inverted signal; and an output inverter for inverting the subset of the data bits of the bit-inverted signal, to regenerate the input one-bit digital signal.

Storage and transmission of one-bit data

Data transmission apparatus for multiplexing one or more channel data streams of equal, predetermined, data rate into a transmission data stream comprises: means for generating a control data stream of the same data rate as the channel data rate, in which first periodically repeating portions of the control data stream provide a synchronisation signal and second periodically repeating portions of the control data stream provide identification data specifying information about the channel data streams with reference to a predetermined ordering of the channel data streams; and means for multiplexing the control data stream with the one or more channel data streams in a cyclic pattern of equally-sized data portions so that a data portion from the control data stream is followed by data portions from each of the channel data streams according to the predetermined ordering of channel data streams.

Multiplexing digital signals

A signal processor for 1-bit signals comprises a fifth order Delta-Sigma Modulator (DSM) having an input for receiving a 1-bit signal and an output at which a processed 1-bit signal is produced by a quantizer. The quantizer receives a p-bit signal from a series of five signal integration stages. Each stage comprises a first 1-bit multiplier coupled to the input, a second 1-bit multiplier coupled to the output, an adder which sums the outputs of the coefficient multipliers and an integrator which integrates the output of the adder. A final stage comprises a coefficient multiplier and an adder. The adder sums the output of the coefficient multiplier and the output of the integrator of the preceding integration stage. The coefficients are chosen to provide an overall attenuation of the input signal and of the quantization noise produced by the quantizer. Reduction of quantization noise allows several DSMs to be connected in series by preventing the build-up of excess noise, which may compromise stability.

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Oxford, United Kingdom

Peter Charles Eastty