Abstract: Multiple channels of audio are combined either to a monophonic composite signal or to multiple channels of audio along with related auxiliary information from which multiple channels of audio are reconstructed, including improved downmixing of multiple audio channels to a monophonic audio signal or to multiple audio channels and improved decorrelation of multiple audio channels derived from a monophonic audio channel or from multiple audio channels. Aspects of the disclosed invention are usable in audio encoders, decoders, encode/decode systems, downmixers, upmixers, and decorrelators.

Abstract: Multiple channels of audio are combined either to a monophonic composite signal or to multiple channels of audio along with related auxiliary information from which multiple channels of audio are reconstructed, including improved downmixing of multiple audio channels to a monophonic audio signal or to multiple audio channels and improved decorrelation of multiple audio channels derived from a monophonic audio channel or from multiple audio channels. Aspects of the disclosed invention are usable in audio encoders, decoders, encode/decode systems, downmixers, upmixers, and decorrelators.

Abstract: Using an M:N variable matrix, M audio input signals, each associated with a direction, are translated to N audio output signals, each associated with a direction, wherein N is larger than M, M is two or more and N is a positive integer equal to three or more. The variable matrix is controlled in response to measures of: (1) the relative levels of the input signals, and (2) the cross-correlation of the input signals so that a soundfield generated by the output signals has a compact sound image in the nominal ongoing primary direction of the input signals when the input signals are highly correlated, the image spreading from compact to broad as the correlation decreases and progressively splitting into multiple compact sound images, each in a direction associated with an input signal, as the correlation continues to decrease to highly uncorrelated.

Abstract: A hybrid stereophonic/monophonic audio signal encoding comprises generating, in response to a discrete two-channel stereophonic audio signal, an encoded hybrid stereophonic/monophonic audio signal in which the audio signal is a discrete two-channel audio signal below a frequency fm and a single-channel monophonic audio signal above the frequency fm, generating, in response to the discrete two-channel stereophonic audio signal, spatial parameter information characterizing the discrete two-channel stereophonic audio signal above the frequency fm, and combining the hybrid stereophonic/monophonic audio signal with said spatial parameter information in such a manner that the resulting signal is decodable both by a decoder configured to decode a discrete two-channel stereophonic audio signal encoded with the same encoding as applied to the hybrid stereophonic/monophonic audio signal and by a decoder configured to decode, with the use of the spatial parameter information, the hybrid stereophonic/monophonic audio sig

Abstract: A process for translating M audio input channels representing a soundfield to N audio output channels representing the same soundfield, wherein each channel is a single audio stream representing audio arriving from a direction, M and N are positive whole integers, and M is at least 2, generates one or more sets of output channels, each set having one or more output channels. Each set is associated with two or more spatially adjacent input channels and each output channel in a set is generated by a process that includes determining a measure of the correlation of the two or more input channels and the level interrelationships of the two or more input channels.

Abstract: An audio crosstalk-cancelling network that may be implemented in software, such that when run in real time on a personal computer, the canceller has very low mips requirements and uses a small fraction of available CPU cycles. The network is particularly useful for rendering surround sound images outside the space between left and right computer multimedia loudspeakers when the audio from such sources is reproduced. The network includes two signal feedback paths, each feedback path having a time delay and frequency dependent characteristic. The frequency dependent characteristic represents the smoothed difference in the attenuation in the acoustic path between a transducer and the listener's ear farthest from said transducer and the attenuation in the acoustic path between the same transducer and the listener's ear closest to said same transducer. The smoothed difference in the attenuation is implemented by one or more simple digital filters requiring low processing power.

Abstract: Any of several information processing techniques may be used in various information storage and transmission applications to prevent the occurrence of certain “forbidden” bit patterns. According to an encoding technique, a reversible coding process is used to generate an encoded representation of an information stream that cannot contain any forbidden data patterns. This may be accomplished by partitioning the information stream into segments and encoding each segment according to a respective encoding key that is selected such that the results of the coding process cannot contain a forbidden data pattern. According to one substitution technique, all occurrences of forbidden data patterns are replaced with permissible data patterns that do not otherwise occur in the information stream.

Abstract: In an encoder, multiple channels of audio information representing multidimensional sound fields are split into subband signals and the subband signals in one or more subbands are combined to form composite signals. The composite signals, the subband signals not combined into a composite signal and information describing the spectral levels of subband signals combined into composite signals are assembled into an encoded output signal. The spectral level information conveys either the amplitude or power of the combined subband signals or the apparent direction of the sound field represented by the combined subband signals. In digital implementations, adaptive bit allocation may be used to reduce the informational requirements of the encoded signal.

Abstract: In an encoder, multiple channels of audio information representing multidimensional sound fields are split into subband signals and the subband signals in one or more subbands are combined to form composite signals. The composite signals, the subband signals not combined into a composite signal and information describing the spectral levels of subband signals combined into composite signals are assembled into an encoded output signal. The spectral level information conveys either the amplitude or power of the combined subband signals or the apparent direction of the sound field represented by the combined subband signals. In digital implementations, adaptive bit allocation may be used to reduce the informational requirements of the encoded signal.

Abstract: A split-band coding system combines multiple channels of input signals into various forms of composite signals and generates spatial-characteristic signals representing soundfield spatial characteristics in a plurality of frequency subbands. The spatial-characteristics signals may be generated in either or both of two forms. In a first form, the signal represents measures of signal levels for subband signals from the input channels. In a second form, the signal represents one or more apparent directions for the soundfield. The type of the spatial-characteristics signal may be adapted dynamically in response to a variety of criteria including input signal characteristics. Temporal smoothing and spectral smoothing of the spatial-characteristics signals may be applied in an encoder. Temporal smoothing and spectral smoothing may be applied to gain factors derived from the spatial-characteristics signals in a decoder.

Abstract: A multi-channel signal compressor for compressing digital sound signals in the respective channels of a multi-channel sound system. The apparatus comprises a first-stage compression system and a second-stage compression system. In the first-stage compression system, a coupling circuit performs coupling between the digital sound signals of at least two of the channels to generate coupling-processed signals, one for each of the channels. A compressor circuit receives the coupling-processed signals from the coupling circuit and frequency divides each coupling-processed signal into frequency range signals in respective frequency ranges, and compresses the frequency range signals obtained by dividing each coupling-processed signal to generate a first-stage compressed signal.

Abstract: A surround sound encoder, intended for implementation in software, runs in real time on a personal computer using low mips and a small fraction of available CPU cycles. In the principal application for the encoder, the Lt and Rt signals of the encoder are mixed with the Lt and Rt signals of a pre-recorded source (e.g., computer game soundtrack, CD ROM, Internet audio, etc.). Alternatively, the encoder may be used by itself or with one or more other virtual encoders to provide a totally user-generated soundfield. The encoder is implemented in either of two ways: the signal being encoded may be panned to one or more of the four inputs of a surround-sound fixed matrix encoder or the signal may be encoded by applying the signal to a surround-sound variable-matrix encoder.

Abstract: A split-band perceptual coding system utilizes generalized waveform predictive coding in frequency bands to further reduce coded signal information requirements. The system including frequency subband each having a bandwidth commensurate with or less than a corresponding critical band of human perception. The order of the predictors are selected to balance requirements for prediction accuracy and rapid response time. Predictive coding may be adaptively inhibited in a band during intervals in which no predictive coding gain is realized.