Abstract: Provided is a method and computer program product for producing an enhanced audio signal for an output device from audio signals received by 2 or more microphones in close proximity to each other. For example, one embodiment of the present invention comprises the steps of receiving a first input audio signal from the first microphone, digitizing the first input audio signal to produce a first digitized audio input signal, receiving a second input audio input signal from the second microphone, digitizing the second input audio input signal to produce a second digitized audio input signal, using the first digitized audio input signal as a reference signal to an adaptive prediction filter, using the second digitized audio input signal as input to said adaptive prediction filter and finally adding a prediction result signal from the adaptive prediction filter to the first digitized audio input signal to produce the enhanced audio signal.

Abstract: Provided is a method, non-transitory computer program product and system for an improved noise suppression technique for speech enhancement. It operates on speech signals from a single or multiple input sources. Background noise monitoring is performed with one or multiple input speech signals to determine if the input speech contains active voice. If the absence of active voice is detected, the spectrum of the input speech is used to update a long-term noise spectrum estimate. In addition, the input from one or more secondary microphones can be used to update a short-term noise spectrum estimate. The input speech spectrum is then compared to the long-term and/or short-term noise spectra, and a selective spectrum gain based shaping is applied to the input speech spectrum to reduce noise.

Abstract: Provided is a method, non-transitory computer program product and system for an improved noise suppression technique for speech enhancement. It operates on speech signals from a single source such as either the output from a single microphone or the reconstructed speech signal at the receiving end of a communication application. The system performs background noise monitoring of an in-coming speech signal and determines its level, and performs a time domain gain calculation. The noise suppressed output signal is the gain shaped original speech signal.

Abstract: Provided is an improved method for source-independent sound field rotation for virtual and augmented reality applications. The method provides sound field rotation for any type of audio recording source without requiring special processing for different audio formats. The method provides an effective approach to rotate any audio source while preserving directional sound source information.

Abstract: Provided is a method, non-transitory computer program product and system for an improved noise suppression technique for speech enhancement. It operates on speech signals from a single source such as either the output from a single microphone or the reconstructed speech signal at the receiving end of a communication application. The system performs background noise monitoring of an in-coming speech signal and determines its level, and performs a time domain gain calculation. The noise suppressed output signal is the gain shaped original speech signal.

Abstract: Provided is a system, method, and computer program product for improving the quality of speech reproduction in wireless applications where the received speech frames are subject to transmission and packet losses. The speech decoding process is dynamically delayed by at least one frame period in order to perform additional error correction and concealment techniques during times when the wireless link quality if below a predetermined threshold. The wireless link is monitored and if the link quality falls below a predetermined threshold, the decoding process is delayed by at least one frame period so that one or more error correcting techniques can be performed to increase the quality of the reconstructed speech.

Abstract: Provided is a system, method, and computer program product for improving the quality of voice communications on a mobile handset device by dynamically and adaptively selecting adjusting the latency of a voice call to accommodate an optimal speech enhancement technique in accordance with the current ambient noise level. The system, method and computer program product improves the quality of a voice call transmitted over a wireless link to a communication device dynamically increasing the latency of the voice call when the ambient noise level is above a predetermined threshold in order to use a more robust high-latency voice enhancement technique and by dynamically decreasing the latency of the voice call when the ambient noise level is below a predetermined threshold to use the low-latency voice enhancement techniques. The latency periods are adjusted by adding or deleting voice samples during periods of unvoiced activity.

Abstract: Provided is a system for adaptively enhancing an end-user's perceived quality, or quality of experience (QoE), of speech and other audio under ambient noise conditions. The system comprises the steps of determining the ambient noise characteristics on a continuous basis to capture the time varying nature of ambient noises, and adaptively determining the most optimal signal shaping to be applied to the audio/speech signal to produce the most appropriate enhancement to compensate for the ambient noise impairment. The system also comprises a signal shaping technique by using an infinite impulse response (IIR) filter that performs the signal modification with a low delay; a multi-level automatic gain control (AGC); and a controlled amplitude clipping module that assures samples are below a certain limit; and outputs the modified signal for playback through a loudspeaker or the like.

Abstract: Provided is a method for adaptively enhancing an end-user's perceived quality, or quality of experience (QoE), of speech and other audio under ambient noise conditions. The method comprises the steps of determining the ambient noise characteristics on a continuous basis to capture the time varying nature of ambient noises, and adaptively determining the most optimal signal shaping to be applied to the audio/speech signal to produce the most appropriate enhancement to compensate for the ambient noise impairment. The method also comprises a signal shaping technique by using an infinite impulse response (IIR) filter that performs the signal modification with a low delay; a multi-level automatic gain control (AGC); and a controlled amplitude clipping module that assures samples are below a certain limit; and outputs the modified signal for playback through a loudspeaker or the like.

Abstract: Provided is a computer program product for adaptively enhancing an end-user's perceived quality, or quality of experience (QoE), of speech and other audio under ambient noise conditions. The computer program product comprises the steps of determining the ambient noise characteristics on a continuous basis to capture the time varying nature of ambient noises, and adaptively determining the most optimal signal shaping to be applied to the audio/speech signal to produce the most appropriate enhancement to compensate for the ambient noise impairment. The computer program product also comprises a signal shaping technique by using an infinite impulse response (IIR) filter that performs the signal modification with a low delay; a multi-level automatic gain control (AGC); and a controlled amplitude clipping module that assures samples are below a certain limit; and outputs the modified signal for playback through a loudspeaker or the like.