Abstract: Speakers do not always operate linearly. Linearity of the speaker can affect the quality of the sound produced by the speaker, i.e., causing distortions in the sound, if the nonlinearites are not accounted for. To determine nonlinearities of the speaker, the speaker is often modeled and measurements are made to estimate the characteristics of the speaker based on the model. By using an angle sensor and a light source, a speaker manager can make a direct measurement of excursion or displacement of the speaker. Moreover, when the angle sensor, the light source, and the light beam are configured appropriately with respect to the moving cone of the speaker, the measurement can be substantially linear with respect to the amount of excursion or displacement. Such measurements are far simpler to use and in some cases more accurate than measurements made by other types of systems.

Abstract: [object] To provide a headphone device in which the influence of individual differences in virtual sound field reproduction is less likely to occur and which may listen external sounds naturally, [solution] A left-side headphone body and a right-side headphone body include speaker arrays which are formed of a plurality of speaker units which are arranged to surround auricles, respectively. The speaker array of the headphone body reproduces a sound field inside a closed curved surface in the vicinity of the auricle using wave field synthesis, and since reverberation or a diffraction effect occurs in the ear of each individual, the influence caused by individual differences is less likely to occur. In addition, the speaker array has the plurality of the speaker units arranged to surround the auricle and is not of a shape that blocks the ear of the listener, and then the external sound can be heard naturally.

Abstract: A multiple speaker array for audio systems. The array comprises multiple range 30 to 40 ohm speakers connected in parallel. The number of speakers are selected to maintain the load on the system amplifier within an acceptable range, usually 1 to 8 ohms. The array may also include a low range speaker such as an 8 ohm sub-woofer. The speaker array “kit” may be used indoors or outdoors and allows the user to distribute an array of speakers and sub woofer over an area to achieve balanced coverage using conventional amplifiers.

Abstract: An automated process for equalizing an audio system and an apparatus for implementing the process. An audio system includes a microphone unit, for receiving the sound waves radiated from a plurality of speakers, acoustic measuring circuitry, for calculating frequency response measurements; a memory, for storing characteristic data of the loudspeaker units and further for storing the frequency response measurements; and equalization calculation circuitry, for calculating an equalization pattern responsive to the digital data and responsive to the characteristic data of the plurality of loudspeaker units. Also described is an automated equalizing system including a acoustic measuring circuitry including a microphone for measuring frequency response at a plurality of locations; a memory, for storing the frequency responses at the plurality of locations; and equalization calculation circuitry, for calculating, from the frequency responses, an optimized equalization pattern.

Abstract: A method of outputting audio in a teleconferencing environment includes receiving audio streams, processing the audio streams according to information regarding effective spatial positions, and outputting, by at least three speakers arranged in more than one dimension, the audio streams having been processed. The information regarding the plurality of effective spatial positions corresponds to a perceived spatial scene that extends beyond the speakers in at least two dimensions. In this manner, participants in the teleconference perceive the audio from the remote participants as originating at different positions in the teleconference room.

Abstract: A depth processing system can employ stereo speakers to achieve immersive effects. The depth processing system can advantageously manipulate phase and/or amplitude information to render audio along a listener's median plane, thereby rendering audio along varying depths. In one embodiment, the depth processing system analyzes left and right stereo input signals to infer depth, which may change over time. The depth processing system can then vary the phase and/or amplitude decorrelation between the audio signals over time to enhance the sense of depth already present in the audio signals, thereby creating an immersive depth effect.

Abstract: A speaker producing total sound immersion by directing the sound in off-horizontal directions, and directing sound in different frequency ranges in different directions. The low range driver, or woofer, is oriented downward, and fitted with a dispersion plate, which deflects the sound along the floor and up into the room. The high range drivers, or tweeters, are oriented upward and set for wide dispersion, so that the high pitched sound is reflected off the ceiling and is more evenly distributed. One midrange driver is set in each side of the speaker box, and the sides are angled, so that the box is in the form of a truncated pyramid. This causes the midrange sound to be reflected off the walls and ceiling of the room. The result is one speaker producing a total sound immersion effect.

Abstract: A multi-channel input signal having at least three original channels is represented by a parameter representation of the multi-channel signal. A first balance parameter, a first coherence parameter, or a first inter-channel time difference between a first channel pair and a second balance parameter, or a second coherence parameter, or a second inter-channel time difference parameter between a second channel pair are calculated. This set of parameters is the parameter representation of the original signals. The first channel pair has two channels, which are different from two channels of a second channel pair. Furthermore, each channel of the two channel pairs is one of the original channels, or a weighted combination of the original channels, and the first channel pair and the second channel pair include information on the three original channels.

Abstract: A method, apparatus, and system for measuring and analyzing the effects of dynamics modifying processors on a signal. This new approach utilizes statistical analysis techniques to provide a direct comparison and evaluation between the processed signal and the unprocessed signal's dynamic characteristics. The method identifies and quantifies Effective Dynamic Range, Clip Tolerance, Lower Limit Tolerance, Crest Factor, and Diminuendo Factor, using either peak or r.m.s values. In an alternative embodiment, the invention allows for user adjustment and control of the relative relationship of Crest Factor and Diminuendo Factor, which the user may perceive as loudness.

Abstract: The present disclosure describes a networked amplifier in an audio video distribution system that converts a digital audio signal to and from an analog audio signal used in a video and audio system that uses a TCP/IP LAN (local area network) using TCP/IP as the communication protocol. The networked amplifier includes a network controller where the controller includes an embedded controller with a processor, memory, and storage functioning as a web server. Further included is a digital signal processor (DSP) that includes a pre-amplifier, amplifier, and an electronic crossover for frequency division. Also included are analog sensors that used for receiving and or gathering control information that enables internal auto adjustment. And further included is an analog driver circuitry.

Abstract: A method performed by an audio decoder for reconstructing N audio channels from an audio signal containing M audio channels is disclosed. The method includes receiving a bitstream containing an encoded audio signal having M audio channels and a set of spatial parameters, the set of spatial parameters including an inter-channel intensity difference parameter and an inter-channel coherence parameter. The encoded audio bitstream is then decoded to obtain a decoded frequency domain representation of the M audio channels, and at least a portion of the frequency domain representation is decorrelated with an all-pass filter having a fractional delay. The all-pass filter is attenuated at locations of a transient. A matrixed version of the decorrelated signals are summed with a matrixed version of the decoded frequency domain representation to obtain N audio signals that collectively having N audio channels where M is less than N.

Abstract: A method performed by an audio decoder for reconstructing N audio channels from an audio signal containing M audio channels is disclosed. The method includes receiving a bitstream containing an encoded audio signal having M audio channels and a set of spatial parameters, the set of spatial parameters including an inter-channel intensity difference parameter and an inter-channel coherence parameter. The encoded audio bitstream is then decoded to obtain a decoded frequency domain representation of the M audio channels, and at least a portion of the frequency domain representation is decorrelated with an all-pass filter having a fractional delay. The all-pass filter is attenuated at locations of a transient. A matrixed version of the decorrelated signals are summed with a matrixed version of the decoded frequency domain representation to obtain N audio signals that collectively having N audio channels where M is less than N.

Abstract: A method performed by an audio decoder for reconstructing N audio channels from an audio signal containing M audio channels is disclosed. The method includes receiving a bitstream containing an encoded audio signal having M audio channels and a set of spatial parameters, the set of spatial parameters including an inter-channel intensity difference parameter and an inter-channel coherence parameter. The encoded audio bitstream is then decoded to obtain a decoded frequency domain representation of the M audio channels, and at least a portion of the frequency domain representation is decorrelated with an all-pass filter having a fractional delay. The all-pass filter is attenuated at locations of a transient. A matrixed version of the decorrelated signals are summed with a matrixed version of the decoded frequency domain representation to obtain N audio signals that collectively having N audio channels where M is less than N.

Abstract: The invention is a multi-channel loudspeaker system that provides a compact loudspeaker configuration and filter design methodology that operates in the digital signal processing domain. Further, the loudspeaker system can be designed to include drivers of various physical dimensions and can achieve prescribed constant directivity over a large area in both the vertical and horizontal planes.

Abstract: Monaurally-recorded mono or stereo recordings may be processed and converted into binaurally-recorded audio recordings. An analog process of performing this involves output of at least subsets of the monaurally-recorded recording, such as isolated instrument/vocal tracks, to be played to a dummy with two microphones. A digital process of performing this includes simulating audio input from simulated locations corresponding to audio sources. A neural network process of performing this includes training a neural network using speakers and microphones and then automating conversion from monaural audio to binaural audio based on the training of the neural network. The neural network can also be trained with output speakers to eliminate or reduce dead zones and/or speaker crosstalk.

Abstract: A sound reproduction system includes an electro-acoustic transducer and a transducer driver for driving the electro-acoustic transducer. The transducer drive includes a filter which is configured to reproduce at a listener's location an approximation to the local sound field that would be present at the listener's ears in recording space, taking into account the characteristics and intended position of the electro-acoustic transducer relative to the listener's ears. The electro-acoustic transducer includes a first sound emitter which provides an intermediate sound emission channel, and second and third sound emitters providing respective left and right sound emission channels. The first sound emitter is located intermediate of second and third sound emitters. Higher frequencies from at least one of the second and third sound emitters are transmitted closer to the first sound emitter while lower frequencies are transmitted away from the first sound emitter.

Abstract: In an active noise reducing headphone, a signal processor applies filters and control gains of both the feed-forward and feedback active noise cancellation signal paths. The signal processor is configured to apply first feed-forward filters to the feed-forward signal path and apply first feedback filters to the feedback signal path during a first operating mode providing effective cancellation of ambient sound, and to apply second feed-forward filters to the feed-forward signal path during a second operating mode providing active hear-through of ambient sounds with ambient naturalness.

Abstract: A speaker system includes an electronic signal processing unit configured to split an input audio signal into a low-pass audio component and a high-pass audio component. The electrical signal processing unit includes a multi-rate finite impulse response (FIR) filter configured to downsample the low-pass audio component and sample the downsampled low-pass audio component at a first sampling rate lower than a second sampling rate used to sample the high-pass audio component. The speaker system further includes an electronic adaptive equalizer coefficient (EAEC) module configured to dynamically determine at least one equalizer coefficient based on a signal characteristic of the input audio signal. An electronic filter coefficient update unit (FCUU) is included and is configured to control a frequency response of the speaker system based on the at least one equalizer coefficient.

Abstract: A system for processing audio data comprising an amplifier configured to receive an audio signal and to perform nonlinear processing on the audio signal. An encoder coupled to the amplifier, the encoder configured to receive the nonlinearly processed audio signal and to encode the nonlinearly processed audio signal into a data transmission format. A transmitter configured to receive and transmit the encoded nonlinearly processed audio signal. A receiver configured to receive the transmitted encoded nonlinearly processed audio signal and to decode the encoded nonlinearly processed audio signal. A digital voice processor configured to receive the nonlinearly processed audio signal and to use the nonlinearly processed audio signal for echo estimation and to subsequently subtract the estimated echo signal from a microphone signal.

Abstract: According to an embodiment, a transducer system includes a transducing element and a symmetry detection circuit. The transducing element includes a signal plate, a first sensing plate, and a second sensing plate. The symmetry detection circuit is coupled to a differential output of the transducer element and is configured to output an error signal based on asymmetry in the differential output.