Abstract: Systems and methods of improved noise reduction using direction of arrival information include: receiving audio signals from two or more acoustic sensors; applying a beamformer module to the audio signals to employ a first noise cancellation algorithm to the audio signals and combine the audio signals into an audio signal; applying a noise reduction post-filter module to the audio signal, the application of which includes: estimating a current noise spectrum of the audio signals after the application of the first noise cancellation algorithm; using spatial information derived from the audio signals received from the two or more acoustic sensors to determine a measured direction-of-arrival by estimating the current time-delay between the acoustic sensor inputs; comparing the measured direction-of-arrival to a target direction-of-arrival; applying a second noise reduction algorithm to the audio signal; and outputting a single audio stream with reduced background noise.

Abstract: Systems and methods of improved noise reduction include the steps of: receiving an audio signal from two or more acoustic sensors; applying a beamformer to employ a first noise cancellation algorithm; applying a noise reduction post-filter module to the audio signal including: estimating a current noise spectrum of the received audio signal after the application of the first noise cancellation algorithm, wherein the current noise spectrum is estimated using the audio signal received by the second acoustic sensor; determining a punished noise spectrum using the time-average level difference between the audio signal received by the first acoustic sensor and the current noise spectrum; determining a final noise estimate by subtracting the punished noise spectrum from the current noise spectrum; and applying a second noise reduction algorithm to the audio signal received by the first acoustic sensor using the final noise estimate; and outputting an audio stream with reduced background noise.

Abstract: Systems and methods of improved noise reduction using direction of arrival information include: receiving an audio signal from two or more acoustic sensors; applying a beamformer module to employ a first noise cancellation algorithm to the audio signal; applying a noise reduction post-filter module to the audio signal, the application of which includes: estimating a current noise spectrum of the received audio signal after the application of the first noise cancellation algorithm; using spatial information derived from the audio signal received from the two or more acoustic sensors to determine a measured direction-of-arrival by estimating the current time-delay between the acoustic sensor inputs; comparing the measured direction-of-arrival to a target direction-of-arrival; applying a second noise reduction algorithm to the audio signal in proportion to the difference between the measured direction-of-arrival and the target direction-of-arrival; and outputting a single audio stream with reduced background noi

Abstract: In one representative embodiment, one or several mono audio signals and corresponding positional information are processed by respective mixing blocks. The positional information may include azimuth information, elevation information, and range information. For each mono audio signal, a plurality of left and right signals are generated using mixing techniques. The corresponding left and right signals from the plurality of mixing blocks are combined resulting in a single set of left and right signals These signals are then processed with the various placement filters and the outputs of the placement filters are combined to generate a stereo signal for output using narrow geometry speakers.

Abstract: The inventive mechanism uses several sub-systems to generate outputs from the stereo input signal. A first sub-system synthesizes the phantom center output, which places the monaural center image between the left and right speakers in front of the listener. A second sub-system synthesizes the virtual surround (or rear) output signals, which places the sound images to the sides of the listener. A third sub-system synthesizes the left and right stereo outputs, and expands the locations of the left and right sound images.

Abstract: The apparent location of sound signals as perceived by a person listening to the sound signals over headphones can be positioned or moved in azimuth, elevation and range by a range control block and a location control block. Several range control blocks and location control blocks can be provided depending on the number of input sound signals to be positioned or moved. All of the range and location control is provided by the range control blocks and location control blocks so that the resultant signals require only a fixed number of filters regardless of the number of input audio signals to provide the signal processing. Such signal processing resulting in accurate positioning and moving of the sound source is accomplished using front and back early reflection filters, left and right reverberation filters, front and back azimuth placement filters having a head related transfer function, and up and down elevation placement filters.

Abstract: Input sound signals are accepted for presentation to a plurality of speakers where the number of physically present speakers is less than the number of input sound signals. The accepted signals are expanded and summed in a manner such that a listener perceives sounds as coming from the same number of speakers as there are input signals while having less speakers actually operational. In one embodiment there is shown a five input system having only two speakers. The system is functional when the inputs are stereo or monaural and when there are all five speakers available.

Abstract: The inventive mechanism produces multiple output signals from a two-channel stereo input signal. The mechanism produces a first pair of output signals which retain the monaural information in the stereo input signal. The first pair of signals each comprise a combination of one of the input channels and filtered signal produced from a difference of both input channels. The mechanism produces a second pair of output signals which lacks the monaural information. The second pair of output signals each comprise a combination of one of the input channels and an inverse filtered signal produced from a difference of both input channels. Q-filters are used to provide the various filtered signals.

Abstract: An apparent location of a sound source is controlled in azimuth and range to a listener of the sound using headphones by a range control block that has variable amplitude scalers and a time delay and by an azimuth control block that also has variable amplitude scalers and time delays. An input audio signal is fed in to the range control block and the values of the scalers and the taps on the delay buffers are read out of look-up tables in a controller that is addressed by an azimuth index value corresponding to any location on a circle surrounding the headphone wearer. Several range control blocks and azimuth control blocks can be provided depending on the number of input audio signals to be located. All of the range and azimuth control is provided by the range control blocks and azimuth control blocks so that the resultant signals require only a fixed number of filters regardless of the number of input audio signals to provide the signal processing.

Abstract: A computer network teleconferencing system is provided in which audio signals are coupled to or associated with source indicators, such as source addresses in TCP protocol packets. Three-dimensional spatialization transforms are applied to audio data according to the source of such data so that, upon reproduction, a listener perceives different participants as being located in different spaced-apart locations within a 3-D spatialization region. Preferably, the user may select or modify the apparent location of the teleconference participants.

Abstract: Monaural inputs are accepted for presentation to a pair of speakers. The output sound image from that pair of speakers is such that a listener will perceive that the sound is stereo and that certain sounds come from a non-existent center speaker and that other sounds come from locations beyond the physical boundaries of the two speakers. The system operates to preserve full tone range at both speakers and does not split the sound from a particular instrument between both speakers. The system can be used to provide sound signals to the rear speakers of a multiple speaker system when the sound input to the rear speakers is either monaural or stereo.

Abstract: A method and apparatus to enhance or improve the sound localization properties of hearing aids determines the interaural level differences required for localizing sounds to a specific point of origin. These determined interaural level differences are then used in the adjusting and fitting of the hearing aids. A number of hearing tests at different frequencies with a sound source that apparently moves relative to the test subject are performed, and by adjusting the response of the hearing aid in the left ear relative to the right ear, for example by attenuating the left ear, the amount of attenuation or gain can be found that causes the test subject to perceive that the sound source is located at a specific point of origin. These determined interaural level differences are then used to design a new hearing aid or to modify the existing hearing aid.

Abstract: A system for enhancing the effect of stereo audio signals while maintaining the central audio image includes an expander circuit that tracks the monaural information that is common to both left and right channels and that uses such monaural information to control variable attenuators that influence the inputs to the left and right audio image placement filters.

Abstract: An apparent location of a sound source is controlled in azimuth and range to a listener of the sound using headphones by a range control block that has variable amplitude scalers and a time delay and by an azimuth control block that also has variable amplitude scalers and time delays. An input audio signal is fed in to the range control block and the values of the scalers and the taps on the delay buffers are read out of look-up tables in a controller that is addressed by an azimuth index value corresponding to any location on a circle surrounding the headphone wearer. Several range control blocks and azimuth control blocks can be provided depending on the number of input audio signals to be located. All of the range and azimuth control is provided by the range control blocks and azimuth control blocks so that the resultant signals require only a fixed number of filters regardless of the number of input audio signals to provide the signal processing.

Abstract: The sound field in a stereo reproduction system is enhanced by a preprocessor that removes a portion of the audio information that is common or substantially common to both the left and right stereo input signals before processing the signals in left and right sound placement filters. The left and right placement filter output signals, from which a portion of the common audio information was previously removed before processing, are added to the right and left stereo input signals, respectively, to produce enhanced sound field stereo output signals. The input signals that do not undergo placement processing can be delayed in delay filters to improve coherency when the signals are added and both the placement filters and the delay filters can be implemented by a series of cascaded bi-quadratic filters.

Abstract: A system for processing an audio signal for playback over headphones in which the apparent sound source is located outside of the head of the listener processes the input signal as if it were made up of a direct wave portion, an early reflections portion, and a reverberations portion. The direct wave portion of the signal is processed in filters whose filter coefficients are chosen based upon the desired azimuth of the virtual sound source location. The early reflection portion is passed through a bank of filters connected in parallel whose coefficients are chosen based on each reflection azimuth. The outputs of these filters are passed through scalars to adjust the amplitude to simulate a desired range of the virtual sound source. The reverberation portion is processed without any sound source location information, using a random number generator, for example, and the output is attenuated in an exponential attenuator to be faded out.