Other

Dolby Laboratories Licensing Corporation

Mqa Limited

Malcolm James Law

Peter Graham Craven

John Robert Stuart

Vinay Melkote

Rhonda J Wilson

Roy M Fejgin

Andy Jaspar

Simon Plain

Chris Travis

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

Methods and devices are described for losslessly watermarking an audio signal by performing a noise shaped quantisation and clipping the output from the noise shaped quantisation to bounds computed by a pair of quantised linear functions with gradient 0.5 of the input to the noise shaped quantisation. Corresponding methods and devices are also described for inverting the process to recover an exact replica of the original audio signal.

Transparent lossless audio watermarking enhancement

A method of encoding adaptive audio, comprising receiving N objects and associated spatial metadata that describes the continuing motion of these objects, and partitioning the audio into segments based on the spatial metadata. The method encodes adaptive audio having objects and channel beds by capturing a continuing motion of a number N objects in a time-varying matrix trajectory comprising a sequence of matrices, coding coefficients of the time-varying matrix trajectory in spatial metadata to be transmitted via a high-definition audio format for rendering the adaptive audio through a number M output channels, and segmenting the sequence of matrices into a plurality of sub-segments based on the spatial metadata, wherein the plurality of sub-segments are configured to facilitate coding of one or more characteristics of the adaptive audio.

Audio segmentation based on spatial metadata

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

Methods which uses interpolated primitive matrices to decode encoded audio to recover (losslessly) content of a multichannel audio program and/or to recover at least one downmix of such content, and encoding methods for generating such encoded audio. In some embodiments, a decoder performs interpolation on a set of seed primitive matrices to determine interpolated matrices for use in rendering channels of the program. Other aspects are a system or device configured to implement any embodiment of the method.

Rendering of multichannel audio using interpolated matrices

A method of decomposing a matrix of dimension L-by-N, where L is less than or equal to N, into a sequence of N-by-N unit primitive matrices and a permutation matrix comprising a sequence that is the product of the primitive matrices and the permutation matrix, containing L rows that are substantially close to the provided L-by-N matrix, where the choice of the permutation matrix and the indices of the non-trivial rows in the primitive matrices are chosen to limit the coefficient values in the primitive matrices.

Matrix decomposition for rendering adaptive audio using high definition audio codecs

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.

Microphone array

A lossless encoder and decoder are provided for transmitting a multichannel signal on a medium such as DVD-Audio. The encoder accepts additionally a downmix specification and splits the encoded stream into two substreams, such that a two-channel decoder of meagre computational power can implement the downmix specification by decoding one substream, while a multichannel decoder can decode the original multichannel signal losslessly using both substreams. Further features provide for efficient implementation on 24-bit processors, for confirmation of lossless reproduction to the user, and for benign behaviour in the case of downmix specifications that result in overload. The principle is also extended to mixed-rate signals, where for example some input channels are sampled at 48 kHz and some are sampled at 96 kHz.

Matrix improvements to lossless encoding and decoding

In one aspect of the invention, a stream is divided into two substreams, the first substream providing information relating to a 'downmix' signal obtained by matrixing and containing fewer channels than an original multichannel digital signal, and the second substream providing additional information allowing the original multichannel digital signal to be losslessly recovered by a decoder. In a preferred implementation, an encoder furnishes the downmix signal using a cascade of one or more primitive matrix quantizers, each of which implements an n-by-n matrix, followed by selection of the m channels required for the downnix. In a second aspect, a lossless compression system includes a dither seed in the encoded bitstream. The dither seed is used to synchronize a pseudo-random sequence generator in the decoder with a functionally identical generator in an encoder. In a third aspect, encoders and decoders contain uncommitted primitive matrix quantizers, the encoder having logic that accepts a downmix specified as a matrix of coefficients, allocates a number of primitive matrix quantizers to furnish the downmix and optionally allocates a further number to provide matrixing to reduce the data rate.

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Steyning, United Kingdom

Malcolm James Law