Other

Dolby Laboratories Licensing Corporation

Mqa Limited

B & W Loudspeakers Limited

Lenbrook Industries Limited

Cirrus Logic Inc.

Meridian Audio Limited

Reinet S.a.r.l.

Craven, Peter Graham

Law, Malcolm

Travis, Christopher Julian

Craven, Peter G.

Travis, Chris

Peter Graham Craven

Malcolm James Law

John Robert Stuart

Malcolm Law

Michael Anthony Gerzon

Michael A Gerzon

Christopher Julian Travis

Chris Travis

Peter Herbert Gerzon

Malcolm J Law

Encoding and decoding systems are described for the provision of high quality digital representations of audio signals with particular attention to the correct perceptual rendering of fast transients at modest sample rates. This is achieved by optimising downsampling and upsampling filters to minimise the length of the impulse response while adequately attenuating alias products that have been found perceptually harmful.

Digital encapsulation of audio signals

Methods and devices are described for reducing the audible effect of pre-responses in an audio signal. The pre-responses are effectively delayed by employing a digital non-minimum-phase filter, which includes a zero lying outside the unit circle in its z-transform response. This zero is not paired with another zero at a reciprocal position inside the unit circle, as this would linearise the phase modification. The filtering can introduce a greater group delay at the pre-response frequency than at a low frequency, such as 500 Hz or even 0 Hz. The technique can be used to reduce pre-responses in an existing audio signal and also to pre-empt pre-responses that would be introduced to the audio signal by subsequent processing.

Non linear filter with group delay at pre-response frequency for high res radio

A compact multi-element microphone has two rings of directional sensors. Using simple analog electronics, it delivers first-order outputs with low noise, wide bandwidth and tight transient response. The double-ring structure provides exceptionally high directional fidelity in the horizontal plane, while also keeping out-of-plane behaviour under control. This enables faithful capture of ambience, reflections and reverberation. A non-radial capsule arrangement moderates cavity resonances and reduces shading. Combined with digital electronics, the array can efficiently provide second-order and higher-order horizontal directivities that maintain their performance over a wider frequency range than with prior solutions. Outputs can be mono, two-channel stereo and multichannel surround sound. Applications include 360-degree immersive audio, with-height concert hall recording, and advanced voice capture using electronic steering of beams and nulls.

Microphone arrays providing improved horizontal directivity

Quantisation methods are provided which employ dither techniques to reduce the noise penalty in certain circumstances whilst still removing noise modulation. One method relates to reducing the wordwidth of audio by one bit, while another method relates to burying one bit of data in a pair of signal samples.

Discrete dither

Methods and devices are described for lossless bandsplitting and bandjoining of streams of signal samples using allpass filtering. The bandsplitting operation reformats an original stream into two intermediate streams representing even and odd samples of the original stream, and then matrix filters these to provide two output substreams representing higher frequency components and lower frequency components of the original stream. Conversely, the bandjoining operation matrix filters two subband streams to provide two quantised intermediate substreams, and then interleaves the filtered streams to furnish an output stream, such that the intermediate substreams are the even and odd samples of the output stream.

Lossless bandsplitting and bandjoining using allpass filters

Non linear filter with group delay at pre-response frequency for high res audio

Encoding and decoding systems are described for the provision of high quality digital representations of audio signals with particular attention to the correct perceptual rendering of fast transients at modest sample rates. This is achieved by optimizing downsampling and upsampling filters to minimize the length of the impulse response while adequately attenuating alias products that have been found perceptually harmful.

An encoding method and encoder is provided for transparent lossless audio watermarking by quantizing an original PCM audio signal twice, each quantization quantizing to a quantization grid. As a PCM signal is inherently already quantized, there are three quantization grids to consider, the first being the quantization grid of the original PCM signal, the second being that of the watermarked signal and the third being that of an intermediate signal. The technique reduces the amount of introduced quantization error, spectrally shapes the error and fully decorrelates signal alterations from the original audio, thus making the error more similar to additive noise. A decoding method and decoder is also provided, as is a method of altering the watermark without fully decoding the encoded signal.

Transparent lossless audio watermarking

Methods are disclosed for an encoder to embed a data stream into a quantized PCM digital audio signal and for a corresponding decoder to both retrieve the data stream and losslessly reconstruct the exact original audio. Some methods employ complimentary amplification and attenuation, while others employ gain redistribution. Pre-emphasis and soft clipping techniques are described as methods of losslessly reducing the peak excursion of the PCM audio signal. Also described is the lossless placing of data at predetermined positions within an audio stream.

Lossless embedded additional data

An encoder for digital audio signals at a higher sample rate creates a stream for consumer distribution at a lower sampling rate, with compatibility for standard PCM players without a decoder. In conjunction with a suitable decoder, two enhanced playback options are supported, the first option allowing full lossless reconstruction of a noise-shaped higher sampling rate signal, the second option allowing lossy bandwidth extension even if an intervening transmission chain has truncated the least-significant-bits of the encoder's output signal.

Doubly compatible lossless audio bandwidth extension

In lossy data compression of a signal using ADPCM, an adaptive decorrelation or 'prediction' filter is used to reduce the amplitude of the signal, the spectral dynamic range of the signal also being reduced. This latter reduction is effected in a nonuniform manner, if known techniques are used, with regions of high spectral density being compressed more than regions of low spectral density. The present invention recognises that using a uniform compression ratio results in a better tradeoff between compression and robustness to transmission channel errors. A method is described for obtaining a uniform compression ratio by adjusting coefficients of the decorrelation filter in dependence on coefficients of an adaptive training filter that is fed from the output of the decorrelation filter. A reverse method is also provided along with encoder, decoder and codec implementing the techniques.

Prediction of signals

A method is described for packing variable-length entropy coded data into a fixed rate data stream along with resolution enhancement data, the method providing tightly constrained propagation of transmission channel errors and graceful degradation of signal resolution as entropy-coded data rate increases. An application to a multiband ADPCM audio codec is also described.

Lossy coding of signals

A sound capture device comprises a symmetric microphone array that includes non-radially-oriented directional sensors (). The device typically derives a spherical harmonic representation of the incident sound field, and affords higher signal-to-noise ratios and better directional fidelity than prior arrays, across a wide range of audio frequencies.

Growing community of inventors

Haslemere, United Kingdom

Peter Graham Craven