Abstract: A multidirectional audio decoder using an “adaptive” audio matrix derives at least one of a plurality of output audio signals from two or more directionally-encoded audio input signal streams (S1(?), S2(?), . . . SN(?), wherein ? is the encoded angle of a source audio signal. Each output signal is associated with a principal direction ?. In order to generate each output signal, a pair of intermediate signals (“antidominant” signals) are generated, constituting the antidominant signal for each of the two adjacent principal output directions of the decoder. The antidominant signal for any arbitrary principal (or “dominant”) direction is the combination of input signals having coefficients such that the combination goes to zero for that dominant direction. Amplitude control is applied to the two antidominant signals to deliver a pair of signals having substantially equal magnitudes that are additively or subtractively combined to provide the output audio signal associated with a principal direction.

Abstract: Codeword-position-caused encoder latency is reduced by avoiding the requirement for knowledge of the message prior to generating an error detecting or concealing codeword associated with the message. A pseudo error detecting or concealing codeword is inserted in place of the normal error detecting or concealing codeword appropriate for the segment of information to which the error detecting or concealing codeword relates. In order to satisfy the requirement of conventional decoders, the pseudo error detecting or concealing information must match or be appropriate for the segment so that the decoder sees the codeword and message segment as valid or error free. This is accomplished by modifying or perturbing at least a portion of the segment to which the pseudo codeword relates. The invention is particularly useful for maintaining the backward compatibility of audio data encoding formats in which the minimum latency is too long (e.g.

Abstract: A signal processing system comprising components such as split-band perceptual coders can receive a peak amplitude limited input audio signal and can process the signal in such a manner that the processed signal preserves the apparent loudness of the input signal but is no longer peak-amplitude limited. In one embodiment, upsampling is used in estimating the resultant peak amplitude and gain factors established in response to the estimated peak amplitude are applied to one or more frequency subbands of the processed signal.

Abstract: One-bit analog-to-digital converters (ADCs) and digital-analog-converters (DACs) employ an adaptive filter. The filter has two regimes of operation: variable gain within the passband of the filter under low-level signal conditions and fixed gain but a variable filter cutoff frequency (sliding band) under high-level signal conditions. Thus, excessive low-frequency gain under no-signal conditions is avoided and, when implemented using a voltage controlled amplifier (VCA), the arrangement does not demand less offset from the VCA under low-level signal conditions.

Abstract: The complementary pre-processing and post-processing circuits of this inventiona re advantageous for reducing noise introduced into a signal by a process or medium where the noise introduced rises with the signal level. The pre-processing circuit analyzes the spectral composition of an input signal and generates an emphasis control signal to so indicate. The emphasis control signal is limited by a limitation circuit to a desired bandwidth. The pre-processing circuit includes an emphasis applying circuit for amplifying or attenuating the input signal as a function of the emphasis control signal. A delay circuit delays the input signal relative to the emphasis control signal so that the input signal reaches the emphasis applying circuit when the emphasis control signal is available for controlling the amplification or attenuation applied.The post-processing circuit restores the spectral composition of the signal.

Abstract: An encoder converts an analog input signal to a digital signal. The encoder comprises a circuit for generating a control signal with a predetermined bandwidth from the input analog signal. The control signal indicates the desired optimum step-size to be applied in the encoder. The encoder further comprises a converter for converting the analog input signal to a digital signal. The converter is responsive to the control signal representing step-size so that the digital signal will be generated in accordance with the step-size information. Since the generation of a limited bandwidth control signal requires time, the encoder also includes a circuit for introducing time delay to the analog input signal, so that the delayed input signal reaches the converter when the control signal is available for digital encoding.A decoder generates an analog output signal from a digital signal received from an encoder through a transmission medium.

Abstract: The time-base corrector of this invention corrects the time-base of audio signals reproduced from video tapes without correcting the time-base of video signals which are associated with the audio signals. Reference signals such as the horizontal synchronizing pulses which accompany video information are used for correcting the time-base of audio signals if such reference signals have been subjected to substantially the same time compression or expansion as the audio signals. In the preferred embodiment, the reproduced audio signals are sampled and written into a sampled data delay and read out of it by means of a control circuit. The control circuit receives as input the undistorted reference signal and the reproduced reference signal which has undergone time compression or expansion. The control circuit derives a write clock signal with a frequency equal to k times the frequency of the distorted reference signal.

Abstract: Apparatus for reducing unmasked quantizing noise modulation in adapting differential digital coding systems. In the digital encoder low frequency signals are boosted in the presence of dominating high frequency signals that are controlling the quantizing step size. The decoder has complementary cutting of low frequency signals. Optionally, unmasked high frequency noise modulation is reduced by variable pre- and de-emphasis.

Abstract: In order to reduce the effect of modulation of the quantizing error (noise) by the signal in an adaptive digital audio system and to increase the dynamic range of the system, the adaptive analog to digital converter is preceded by a frequency dependent network that changes the shape of its characteristic in response to a control signal. The control signal is derived from the variable scaling factor in the adaptive converter. After transmission via a transmission channel or record/playback process, complementary operations are performed by a digital to analog converter followed by a further frequency dependent variable response network.

Abstract: A system for recording analog information on a magnetic medium in which the record bias is dynamically varied in response to the amplitude in a predetermined frequency spectrum of the information to be recorded. The record amplifier gain and equalization may be simultaneously dynamically varied as the record bias in order to maintain a record/playback response relatively invariant with respect to changes in the bias level.

Abstract: A network is described which processes two channel stereophonic sound programs for playback through left, right and center sound reproduction channels. Use of the network allows certain advantages of a three channel program to be obtained from the two channel program. The network delays the left and right channels to give precedence to the center channel information but reduces the level of this information to enable the left and right channels to establish a stereo image.