Mindspeed Technologies, Inc.

Conexant Systems, Inc.

Qosound, Inc.

Rockwell International Corporation

Other

Samsung Electronics Co., Ltd.

Intel Corporation

Rockwell Semiconductor Systems, Inc.

Htc Corporation

Huan-Yu Su

Adil Benyassine

Yang Gao

Eyal Shlomot

Jes Thyssen

Anthony Jiming Su

Carlo Murgia

Jonathan Peace

Jeffrey D Klein

Michael M Metzger

Tom Hong Li

Armin Abold

James W Johnston

Patrick D Ryan

Khaled Assaleh

Gilles G Fayad

Nick J Lavrov

Thomas H Eichenberg

Adil Benyassinc

Oleg Khaykin

Eric Kwok Fung Yuen

Marco Y C Cheng

Eric Kwok Yuen

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. In other embodiments, any number of microphones can be used, and in all embodiments there is no requirement to detect or locate the source or direction of arrival of the input audio signals.

Adaptive multi-microphone beamforming

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.

Noise suppressor

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.

Noise supressor

Provided is a system for wireless communications including several base stations supporting a wide area wireless network and several mobile user equipment (UE) devices. Each mobile UE device may be configured to transmit a request to establish a local wireless connection with one or more of the UE devices. The mobile UE device may receive a response containing connectivity information from each of the mobile UE devices and then select one of the mobile UE devices based on the connectivity information received from each of the mobile UE devices. The mobile UE device may then establish a local wireless connection with the selected mobile UE device. The mobile UE device may then communicate with one of the base stations in the wide area wireless network through the selected mobile UE device, utilizing the local wireless connection.

Efficiency of wireless wide area networks utilizing local wireless connections

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.

Source independent sound field rotation for virtual and augmented reality applications

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.

Adaptive delay for enhanced speech processing

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.

Error concealment for speech decoder

There is provided a method of using an adaptive tilt compensation by a speech decoder. The method comprises receiving a bit stream including a plurality of parameters representative of a speech signal; identifying an adaptive code vector and a fixed code vector using the plurality of parameters; scaling the adaptive code vector and the fixed code vector to generate a scaled adaptive code vector and a scaled fixed code vector; summing the scaled adaptive code vector and the scaled fixed code vector to generate a synthesized output; calculating a first reflection coefficient based on the plurality of parameters representative of the speech signal; multiplying the first reflection coefficient by a factor to generate a tilt factor; and applying the tilt factor to the synthesized output based on an encoding bit rate.

Adaptive tilt compensation for synthesized speech

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.

System for adaptive audio signal shaping for improved playback in a noisy environment

There is provided a method of encoding an input speech signal. The method comprises identifying a fixed codebook vector from a fixed codebook; identifying an adaptive codebook vector from a adaptive codebook; calculating an adaptive codebook gain; reducing the adaptive codebook gain by an amount; optimally selecting a fixed codebook gain based on the adaptive codebook gain while both the fixed codebook vector and the adaptive codebook vector remain fixed; and converting the input speech signal into an encoded speech using the fixed codebook gain, the adaptive codebook gain, the fixed codebook vector and the adaptive codebook vector. The amount of reducing the adaptive codebook gain may be varied.

Adaptive gain reduction for encoding a speech signal

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.

Computer program product for adaptive audio signal shaping for improved playback in a noisy environment

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.

Method for adaptive audio signal shaping for improved playback in a noisy environment

A method comprises analyzing each frame of a plurality of frames of the speech signal to determine one or more speech parameters for the speech signal; deciding, for each frame of the plurality of frames of the speech signal, based on the one or more speech parameters of the speech signal, to select one of a plurality of encoding modes including a first encoding mode and a second encoding mode for encoding each frame of the plurality of frames of the speech signal; encoding each frame of the plurality of frames of the speech signal according to the selected one of the plurality of encoding modes for each frame of the plurality of frames in the deciding; the first encoding mode supports a first encoding rate and the second encoding mode supports a second encoding rate, wherein the first encoding rate is the same encoding rate as the encoding rate.

Multi-mode speech encoding system for encoding a speech signal used for selection of one of the speech encoding modes including multiple speech encoding rates

There is provided a method for use by an echo canceller to cancel a far echo at a variable delay and a near echo at a fixed delay. The method comprises constructing an echo signal model based on an incoming signal, determining a variable echo delay for a far echo caused by a far echo source, determining a fixed echo delay for a near echo caused by a near echo source, subtracting the echo signal model from an outgoing signal at a window placed around the variable echo delay to cancel far echo, e.g. when the echo canceller determines existence of the far echo, and subtracting the echo signal model from the outgoing signal at a window placed around the fixed echo delay to cancel near echo, e.g. regardless of existence of the near echo, wherein the fixed echo delay is smaller than the variable echo delay.

Growing community of inventors

San Clemente, CA, United States of America

Huan-Yu Su