Abstract:
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.
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 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.
Abstract:
A signal convertor comprising a pulse modulator, and a modifier for modifying the signal input thereto in dependence upon the error in previous values of the output thereof, to reduce the effects of said error within a desired signal band.
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:
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:
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 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.
Abstract:
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.
Abstract:
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.