Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: An encoding method and encoder is provided for transparent lossless audio watermarking by quantising an original PCM audio signal twice, each quantisation quantising to a quantisation grid. As a PCM signal is inherently already quantised, there are three quantisation grids to consider, the first being the quantisation 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 quantisation 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.

Abstract: 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.

Abstract: Methods and devices are described whereby a representation of an original PCM signal may be reversibly degraded in a controlled manner and information losslessly embedded to produce a streamable PCM signal, which provides a controlled audio quality when played on standard players and conditional access to a lossless presentation of the original PCM signal. Using such techniques allows control over the level of degradation of the signal and also flexibility in the type information of information embedded. Some methods require a song key, which is employed in one or both of the degrading and embedding steps and for creating a token. These methods may further require a user key, which is used to encrypt the song key before creating the token.

Abstract: Methods are disclosed for an encoder to embed a data stream into a quantised 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.

Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: A sound capture device comprises a symmetric microphone array that includes non-radially-oriented directional sensors (101). 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.

Abstract: 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.

Abstract: 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.

Abstract: A sound capture device comprises a symmetric microphone array that includes non-radially-oriented directional sensors (101). 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.

Abstract: 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.