Abstract: Embodiments are described for designing a filter in a magnitude domain performing an impedance filtering function over a frequency domain to compensate for directional cues for the left and right ears of the listener as a function of virtual source angles during headphone virtual sound reproduction. The filter is derived by obtaining blocked ear canal and open ear canal transfer functions for loudspeakers placed in a room, obtaining an open ear canal transfer function for a headphone placed on a listening subject, and dividing the loudspeaker transfer functions by the headphone transfer function to invert a headphone response at the entrance of the ear canal and map the ear canal function from the headphone to free field.

Abstract: In some embodiments, a method for monitoring speakers within an audio playback system (e.g., movie theater) environment. In typical embodiments, the monitoring method assumes that initial characteristics of the speakers (e.g., a room response for each of the speakers) have been determined at an initial time, and relies on one or more microphones positioned in the environment to perform a status check on each of the speakers to identify whether a change to at least one characteristic of any of the speakers has occurred since the initial time. In other embodiments, the method processes data indicative of output of a microphone to monitor audience reaction to an audiovisual program. Other aspects include a system configured (e.g., programmed) to perform any embodiment of the inventive method, and a computer readable medium (e.g., a disc) which stores code for implementing any embodiment of the inventive method.

Abstract: A method and apparatus for enhancing a desired audio signal for delivery through an electroacoustic channel include obtaining a noise estimate attributable to an external disturbance, applying the noise estimate to a dynamic noise compensation (DNC) process to thereby condition the desired audio signal as a function of the spectral characteristics of the noise estimate, applying the noise estimate to an adaptive equalization (AEQ) process to thereby condition the desired audio signal as a function of the electroacoustic response of the electroacoustic channel, and applying the noise estimate to an active noise cancellation (ANC) process configured to generate anti-noise for delivery into the electroacoustic channel.

Abstract: Frequency-domain techniques are used for adaptive equalization that is responsive to spectral magnitude characteristics but not sensitive to phase characteristics of system response. Signal correlation may be used to improve adaptation accuracy when significant levels of ambient sounds are present. A preferred filter implementation uses convolution-based block transforms and cross-fade windows.

Abstract: In some embodiments, a method for monitoring speakers within an audio playback system (e.g., movie theater) environment. In typical embodiments, the monitoring method assumes that initial characteristics of the speakers (e.g., a room response for each of the speakers) have been determined at an initial time, and relies on one or more microphones positioned in the environment to perform a status check on each of the speakers to identify whether a change to at least one characteristic of any of the speakers has occurred since the initial time. In other embodiments, the method processes data indicative of output of a microphone to monitor audience reaction to an audiovisual program. Other aspects include a system configured (e.g., programmed) to perform any embodiment of the inventive method, and a computer readable medium (e.g., a disc) which stores code for implementing any embodiment of the inventive method.

Abstract: Frequency-domain techniques are used for adaptive equalization that is responsive to spectral magnitude characteristics but not sensitive to phase characteristics of system response. Signal correlation may be used to improve adaptation accuracy when significant levels of ambient sounds are present. A preferred filter implementation uses convolution-based block transforms and cross-fade windows.

Abstract: A method and apparatus for enhancing a desired audio signal for delivery through an electroacoustic channel include obtaining a noise estimate attributable to an external disturbance, applying the noise estimate to a dynamic noise compensation (DNC) process to thereby condition the desired audio signal as a function of the spectral characteristics of the noise estimate, applying the noise estimate to an adaptive equalization (AEQ) process to thereby condition the desired audio signal as a function of the electroacoustic response of the electroacoustic channel, and applying the noise estimate to an active noise cancellation (ANC) process configured to generate anti-noise for delivery into the electroacoustic channel.

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: 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 audio applications, a digital split-band encoder splits an input signal into frequency subband signals having bandwidths commensurate with the critical bandwidths of the human auditory system, 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 with allowance for decoding synthesis filter bank spectral distortions.

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: The invention relates to formatting encoded audio information in a form suitable for transmission or storage. Audio information is encoded into a binary form, using an invariant number of bits to represent at least some but not all of the encoded information. The information represented by an invariant number of bits is assembled into pre-established positions within a formatted frame.

Abstract: The invention relates to the design of analysis and synthesis windows for use in high-quality transform encoding and decoding of audio signals, especially encoding and decoding having a short signal-propagation delay. The design method derives a pair of analysis/synthesis windows from a known window function which satisfy various filter selectivity and window overlap-add constraints.

Abstract: A transform encoder, a transform decoder, and a transform encoder/decoder system utilize complex pre- and post-transform multiplication of input signal samples to implement concurrent application of a modified Discrete Cosine Transform and a modified Discrete Sine Transform according to the Evenly-Stacked Time Domain Aliasing Cancellation technique against two channels of input signal samples, and to reduce the computational complexity of a digital filter bank of a modified Discrete Cosine Transform.

Abstract: A transform encoder, a transform decoder, and a transform encoder/decoder system employ adaptive bit allocation wherein each code word representing spectral information is allocated a fixed number of bits and an adaptive number of bits, except that at least some but not all code words are allocated a fixed number of bits.

Abstract: Quantizing noise is lessened in a speech signal system by using adaptive bit allocation for subband channels, wherein subband information of digital words is represented in block-floating-point form, and normalized mantiss as may allow dropping a sign bit.