Abstract: Methods and apparatus are described by which equalization and/or bass management of speakers in a sound reproduction system may be accomplished.

Abstract: Methods for generating a palette of feedback (IIR) filter coefficient sets and using the palette to configure (e.g., adaptively update) a prediction filter which includes a feedback filter, and a system for performing any of the methods. Examples of the system include an encoder, including a prediction filter and configured to encode data indicative of a waveform signal (e.g., samples of an audio signal), and a decoder. In some embodiments, the prediction filter is included in an encoder operable to generate (and assert to a decoder) encoded data including filter coefficient data indicative of the selected IIR coefficient set with which the prediction filter was configured during generation of the encoded data. In some embodiments, the timing with which adaptive updating of prediction filter configuration occurs or is allowed to occur is constrained (e.g., to optimize efficiency of prediction encoding).

Abstract: M audio input channels, each associated with a spatial direction, are translated to N audio output channels, each associated with a spatial direction, wherein M and N are positive whole integers, M is three or more, and N is three or more, by deriving the N audio output channels from the M audio input channels, wherein one or more of the M audio input channels is associated with a spatial direction other than a spatial direction with which any of the N audio output channels is associated, and at least one of the one or more of the M audio input channels is mapped to a respective set of at least three of the N output channels. At least three output channels of a set may be associated with contiguous spatial directions.

Abstract: Ambience signal components are obtained from source audio signals, matrix-decoded signal components are obtained from the source audio signals, and the ambience signal components are controllably combined with the matrix-decoded signal components. Obtaining ambience signal components may include applying at least one decorrelation filter sequence. The same decorrelation filter sequence may be applied to each of the input audio signals or, alternatively, a different decorrelation filter sequence may be applied to each of the input audio signals.

Abstract: An audio encoder/decoder system suitable for digital motion picture film soundtracks provides for adjusting playback signal gain and dynamic range by modifying the scale factors of scaled spectral information. In one embodiment of a transform coding system, transform coefficients are expressed in floating-point form and adjustments are made by modifying the exponents in the floating-point representation.

Abstract: An adaptive high-fidelity perceptual subband encoder for spatially directional multichannel audio signals, using a mixture of individual-channel subband coding with subband-steered composite-channel signal coding that combines the spectral components of the individual channel subbands selected for steering. The composite channel representation includes a subband steering control signal which either conveys the levels of the spectral components from all the steered channels, or represents the apparent directions (net directional vectors) of the spectral components from all the steered channels.

Abstract: The invention relates in general to low bit-rate encoding and decoding of information such as audio information. More particularly, the invention relates to computationally efficient adaptive bit allocation and quantization of encoded information useful in high-quality low bit-rate coding systems.In one embodiment, an audio split-band encoder splits an input signal into frequency subband signals, quantizes the subband signals according to values established by an allocation function, and assembles the quantized subband signals into an encoded signal. The allocation function establishes allocation values in accordance with psychoacoustic principles based upon a masking threshold.

Abstract: An apparatus and method receive and process information, and adapt the operational characteristics of the processing in response to material included in the received information. More particularly, the apparatus and method embody a software-controlled process which adapts or modifies the software in response to the received information. One embodiment is described in connection with a motion picture film which carries a conventional analog SVA soundtrack and a digital soundtrack with a digital representation of software required to properly process the digital soundtrack.

Abstract: Two or more audio channels (i.e.--stereo, 4-channel surround, etc.) are each divided into frequency subbands to be coarsely quantized. An adaptive bit allocation scheme is then applied to subbands which are combined across channels such that equivalent subbands (i.e.--same frequency band) from each channel are grouped together to form a steered subband. The power from each subband is averaged across the channels to form a steered subband level. Bits are conserved by forming a vector in which each channel is represented by the difference between the steered subband level (average) and the actual subband level. Subbands which are not steered are represented by the coarse quantization and are considered unsteered channel subbands. Subbands which are steered are represented by vectors and are considered composite channel subbands. Vectors may be represented using a lookup table and the steered subband levels may be calculated using alternatives such as a peak level rather than an average.

Abstract: The invention relates in general to low bit-rate encoding and decoding of information such as audio information. More particularly, the invention relates to the suppression of quantizing noise which results from low bit-rate encoding and decoding of signal events such as transients. An encoder identifies intervals of an input signal in which a decoder is likely to generate audible artifacts caused by quantizing errors. Indications of these intervals are passed in the encoded signal to the decoder. The decoder uses these indications to eliminate quantizing artifacts by suppressing corresponding intervals in the output signal.

Abstract: The invention relates in general to the quantization of digital signal information. More particularly, the invention relates to the reduction of perceptible effects of quantizing noise by reserving a special quantizing level for low-level signals in encoding and decoding applications.

Abstract: The invention relates to the reproduction of high-fidelity multi-dimensional sound fields intended for human hearing. More particularly, the invention relates to the decoding of signals representing such sound fields delivered by one or more delivery channels, but played back over a number of presentation channels which may differ from the number of delivery channels. In one embodiment, a subband decoder combines spectral information in the frequency domain prior to inverse filtering, thereby incurring implementation costs roughly proportional to the number of presentation channels rather than to the number of delivery channels.

Abstract: The invention relates generally to an apparatus and a method for receiving and processing information, and for adapting the operational characteristics of the processing in response to material included in the received information. More particularly, the invention relates to an apparatus and a method embodying a software-controlled process which adapts or modifies the software in response to the received information. Although the invention has many applications, the invention is more particularly described in connection with a motion picture application in which film stock carries a conventional analog SVA soundtrack and a digital soundtrack with a digital representation of the software required to properly process the digital soundtrack.

Abstract: An automated translation input system to convert Mechanized Translation System (MTS) orders into Recent Change Messages that can be loaded into a program controlled switch without enhancement or changes in the MTS program to provide a significant saving in Switching Control Center (SCC) labor hours. The new system and method permit using the output of the MTS which would otherwise constitute a printed form for inputting the Automated Translation Input System (ATIS) where the forms are analyzed, the associated Recent Changes are analyzed, and the Recent Change Messages are prepared. The thus prepared Recent Change Messages are inspected by the SCC technician who, through a manual option in the ATIS system, gives the command to download the translations directly into the switch. This eliminates the need for key preparation of the Recent Change forms by the SCC technicians and results in an elimination of keying errors and a significant saving in time.

Abstract: In a system in which a low-bit rate encoder and decoder carries matrixed audio signals, an adaptive rematrix rematrixes matrixed signals from an unmodified 4:2 matrix encoder to separate and isolate quiet components from loud ones, thereby avoiding the corruption of quiet signals with the low-bit-rate coding quantization noise of loud signals. The decoder is similarly equipped with a rematrix, which tracks the encoder rematrix and restores the signals to the form required by the unmodified 2:4 matrix decoder. The encoder adaptive rematrix selects the matrix output signals or the amplitude weighted sum and difference of the matrix output signals. The choice of whether the matrix output signals or the sum and difference of the matrix output signals are selected is based on a determination of which results in fewer undesirable artifacts when the output audio signals are recovered in the decoder.

Abstract: The invention relates to the reproduction of high-fidelity multi-dimensional sound fields intended for human hearing. More particularly, the invention relates to the decoding of signals representing such sound fields delivered by one or more delivery channels, but played back over a number of presentation channels which may differ from the number of delivery channels. In a preferred embodiment, a decoder implemented by a discrete digital inverse transform incurs implementation costs roughly proportional to the number of presentation channels.

Abstract: The decoder of this invention decodes at least two channel signals in a directional information system where at least four input signals containing directional information have been encoded into the two or more channel signals. The decoder generates a first control signal substantially proportional to the logarithm of the ratio of the amplitudes of two of the channel signals to detect, as between two of the channel signals, whether the amplitude of one signal dominates that of the other. The decoder also generates a second control signal substantially proportional to the logarithm of the ratio of the amplitudes of the sum and the difference between two of the channel signals to detect the dominant signal in terms of amplitude. The decoder includes a matrix means responsive to the two or more channel signals and the two control signals for generating a number of output signals according to an algorithm.

Abstract: The decoder of this invention decodes at least two channel signals in a directional information system where at least four input signals containing directional information have been encoded into the two or more channel signals. The decoder generates a first control signal substantially proportional to the logarithm of the ratio of the amplitudes of two of the channel signals to detect, as between two of the channel signals, whether the amplitude of one signal dominates that of the other. The decoder also generates a second control signal substantially proportional to the logarithm of the ratio of the amplitudes of the sum and the difference between two of the channel signals to detect the dominant signal in terms of amplitude. The decoder includes a matrix means responsive to the two or more channel signals and the two control signals for generating a number of output signals according to an algorithm.

Abstract: This invention relates in general to a method and apparatus for identifying specific medical conditions from detected stethoscopic sounds. More particularly, the invention relates to the processing of such sounds electronically and the display of the time-variations of the spectral composition of such sounds. The invention is embodied in an accessory package designed to enable a host computer to perform computerized stethoscopic analysis.

Abstract: An adaptive signal weighting system is disclosed for use in transmitting an electrical information signal of a predetermined bandwidth along a signal path. The system can be used to encode or decode the signal. The system comprises filter means disposed in the signal path for varying the gain impressed on the portion of the information signal within a first select spectral region within the preselected bandwidth. The gain is varied in response to and as a function of a first control signal. Means are provided means for generating the first control signal in response to and in accordance with the signal energy of the information signal substantially within at least a part of the first select spectral region.