Abstract:
An example may include detecting, via a controller, one or more microphones and one or more speakers in an area, measuring, via the one or more microphones, an initial frequency response of an audio signal generated by the one or more speakers inside the area and generating an initial room performance rating, comparing the initial frequency response to a target frequency response, creating audio compensation values to apply to the one or more speakers based on the comparison, and applying the audio compensation values to the one or more speakers.
Abstract:
There are provided herein, a method and system for creating a speech/language pathologies classifier, the method comprising: producing a pathological speech repository of pathological speech samples of multiple impairments; computing speech qualities/pathologies, based on data receive from the pathological speech repository; producing a text repository, the text repository comprises multiple known text passages; converting each one of a selection of the text passages from the multiple known text passages, to a speech segment, while introducing to the speech segment one or more of the computed speech pathologies, thereby creating multiple synthetic impaired speech segments; and training a classifier with the multiple synthetic impaired speech segments thereby creating a speech/language pathologies classifier.
Abstract:
Embodiments of the present invention relate to enhancing sound through reverberation matching. In sonic implementations, a first sound recording recorded in a first environment is received. The first sound recording is decomposed to a first clean signal and a first reverb kernel. A second reverb kernel corresponding with a second sound recording recorded in a second environment is accessed, for example, based on a user indication to enhance the first sound recording to sound as though recorded in the second environment. An enhanced sound recording is generated based on the first clean signal and the second reverb kernel. The enhanced sound recording is a modification of the first sound recording to sound as though recorded in the second environment.
Abstract:
A speech processing device includes a processor; and a memory which stores a plurality of instructions, which when executed by the processor, cause the processor to execute: obtaining input speech, detecting a vowel segment contained in the input speech, estimating an accent segment contained in the input speech, calculating a first vowel segment length containing the accent segment and a second vowel segment length excluding the accent segment, and controlling at least one of the first vowel segment length and the second vowel segment length.
Abstract:
A voice enhancement device including an earpiece configured to be positioned in an ear canal of a user. A microcontroller is operatively coupled to the earpiece. The microcontroller is configured to selectively provide at least multitalker babble. An accelerometer is located within the earpiece and operatively coupled to the microcontroller. The accelerometer is configured to detect speech by the user and communicate with the microcontroller to provide the multitalker babble to the earpiece during the detected speech by the user. A method of making the voice enhancement device, and a method for increasing vocal loudness in a patient using the voice enhancement device are also disclosed.
Abstract:
Systems and methods for improving communication over a network are provided. A system for improving communication over a network, comprises a detection module capable of detecting data indicating a problem with a communication between at least two participants communicating via communication devices over the network, a management module capable of analyzing the data to determine whether a participant is dissatisfied with the communication, wherein the management module includes a determining module capable of determining that the participant is dissatisfied, and identifying an event causing the dissatisfaction, and a resolution module capable of providing a solution for eliminating the problem.
Abstract:
Processing a signal includes: receiving data that includes an input signal; filtering the input signal to generate a filtered signal, such that if the input signal includes at least one instance of a nonlinear distortion of a desired signal then the filtered signal includes a signature signal corresponding to the nonlinear distortion, the nonlinear distortion characterized by a time duration that is within a predetermined range; and detecting whether or not the filtered signal includes the signature signal.
Abstract:
An audio input device is provided which can include a number of features. In some embodiments, the audio input device includes a housing, a microphone carried by the housing, and a processor carried by the housing and configured to modify an input sound signal so as to amplify frequencies corresponding to a target human voice and diminish frequencies not corresponding to the target human voice. In another embodiment, an audio input device is configured to treat an auditory gap condition of a user by extending gaps in continuous speech and outputting the modified speech to the user. In another embodiment, the audio input device is configured to treat a dichotic hearing condition of a user. Methods of use are also described.
Abstract:
Provided is an acoustic signal processing device for producing an output sound meeting listener's preferences by adjusting attack sound, reverberation, and noise component. The device includes: an FFT section for transforming an input audio signal from a time-domain to a frequency-domain to calculate a frequency spectrum signal and for generating a first amplitude spectrum signal and a phase spectrum signal; an attack component controller (10) for controlling an attack component of the first amplitude spectrum signal to generate a second amplitude spectrum signal; a reverberation component controller (20) for controlling a reverberation component of the first amplitude spectrum signal to generate a third amplitude spectrum signal; a first adding section (40) for synthesizing the first amplitude spectrum signal, the second amplitude spectrum signal, and the third amplitude spectrum signal to generate a fourth amplitude spectrum signal; and an IFFT section for generating an audio signal transformed from a frequency domain to a time domain based on the fourth amplitude spectrum signal and the phase spectrum signal generated by the FFT section.
Abstract:
Real-time speech output with improved intelligibility are described. One example embodiment includes a device. The device includes a microphone configured to capture one or more frames of unintelligible speech from a user. The device also includes an analog-to-digital converter (ADC) configured to convert the one or more captured frames of unintelligible speech into a digital representation. Additionally, the device includes a computing device. The computing device is configured to receive the digital representation from the ADC. The computing device is also configured to apply a machine-learned model to the digital representation to generate one or more frames with improved intelligibility. Further, the computing device is configured to output the one or more frames with improved intelligibility. In addition, the device includes a digital-to-analog converter (DAC) configured to convert the one or more frames with improved intelligibility into an analog form. Yet further, the device includes a speaker.