Alcatel Lucent

Lucent Technologies Inc.

Alcatel-lucent USA Inc.

Beijing Xiaomi Mobile Software Co., Ltd.

Walter Etter

Theodore Sizer

Katherine Grace August

Abdelkrim Moulehiawy

Rabih Abou-Chakra

Chin S Chuang

Systems and methods for enabling an audio conference are provided. In one aspect, a system includes a transmitting device which receives audio signals representing sounds captured by at least two microphones from participants situated at a first location. A time-of-arrival delay between at least two of the audio signals is calculated and a beam-formed monaural audio signal and corresponding spatial data are generated and transmitted to remote participants situated at a second location. A receiving device processes the beam-formed monaural audio signal based on the spatial data to render and output spatial audio via speakers to the remote participants at the second location. In various aspects, the spatial data may include an angular value or a participant identifier that are determined based on the time-of-arrival delay. The spatial data may also indicate a total number of conference participants that are detected at the first location.

Systems and methods for audio conferencing

Methods and apparatus are provided for acoustic echo cancellation in a speech signal. Acoustic echo is cancelled by inserting at least one tone in the speech signal, wherein the at least one tone is substantially inaudible to a listener; determining a clock skew between two sampling clocks based on a frequency shift of the at least one tone; re-sampling the speech signal based on the determined clock skew; and performing the acoustic echo cancellation using the re-sampled speech signal. The provided acoustic echo cancellers can be implemented, for example, as terminal-based and/or network-based acoustic echo cancellers. The tone optionally comprises an inaudible tone or multiple tones. The tone generation can be limited to only when a speech power in the vicinity of the tone frequency is larger than a pre-determined threshold, or to the beginning of a call. A level of the tone can optionally be controlled so that the tone is masked by the speech signal.

Clock skew compensation for acoustic echo cancellers using inaudible tones

An exemplary method of facilitating communication includes determining a position of a portable communication device that generates a video output. A sound output control is provided to an audio device that is distinct from the portable communication device for directing a sound output from the audio device based on the determined position of the portable communication device.

Facilitating communications using a portable communication device and directed sound output

A method and apparatus for enabling an improved experience by better matching of the auditory space to the visual space in video viewing applications such as those that may be used in video teleconferencing systems using window-based displays. In particular, in accordance with certain illustrative embodiments of the present invention, one or more desired sound source locations are determined based on a location of a window in a video teleconference display device (which may, for example, comprise the image of a teleconference participant within the given window), and a plurality of audio signals which accurately locate the sound sources at the desired sound source locations (based on the location of the given window in the display) are advantageously generated.

Method and apparatus for improved matching of auditory space to visual space in video teleconferencing applications using window-based displays

A method and apparatus for performing speech quality assessment in a speech communication system (such as, for example, a VoIP communication system) which detects and measures the presence of impulsive noise is provided. Specifically, in one illustrative embodiment, an autoregressive (AR) model of speech (and, in particular, of the excitation of the vocal tract) is advantageously employed to estimate a short-term variance of the speech excitation, and the standard deviation of the speech excitation (i.e., the square root of the variance) is then advantageously compared to a predetermined threshold to identify whether impulsive noise is present. Then, based on a statistic analysis of any such identified impulsive noise, a speech quality assessment is generated.

Method and apparatus for the detection of impulsive noise in transmitted speech signals for use in speech quality assessment

The sidetone noise level in a wireless or wired telephone, for example, is reduced by adjusting both noise and signal levels and/or by improving the signal-to-noise ratio of such signals.

Methods and devices for reducing sidetone noise levels

Enhanced estimation of the noise component of a signal is accomplished by using a plurality of filters. Each filter provides an estimate of a minimum sample in a sample set that includes a plurality of signal samples. A comparator, coupled to the plurality of filters, successively compares the estimates among the plurality of filters, and selects the signal estimate having the lowest magnitude. The selected signal estimate represents an enhanced estimate of the noise component of the signal.

Estimating the noise components of a signal during periods of speech activity

A method and apparatus for modifying the level of a speech signal is provided in which the gain parameter in the encoded speech signal is changed in a variable and cyclical manner over time. More specifically, a speech signal is encoded as a bit stream and the speech signal is transported in frames with each frame potentially being further sub-divided into sub-frames. The gain parameter, e.g., fixed codebook gain, is modified in the speech signal in a variable and cyclical manner over a plurality of frames or sub-frames so that gain is temporally dispersed over a plurality of frames or sub-frames. In effect, the change in the amount of gain applied to the signal is effectively dispersed over time so that gradual and accurate changes in the output level of the signal can be achieved to better match actual signal conditions.

Method and apparatus for adjusting the level of a speech signal in its encoded format

Time-scaled, sound signals (i.e. sounds output at differing speeds) are generated by mixing weighted time-and frequency-domain processed signals, the former signal generally representing speech-based signals while the latter representing music-based signals. The weights applied to each type of signal may be determined by a scaling factor, which in turn is related to the desired speed at which a listener desires to hear a sound signal. In one example of the invention, only stationary signal portions of an input sound signal are used to generate time-scaled processed signals. An adaptive frame-size may also be used to pre-process the separate signals prior to being weighted, which at least decreases the amount of unwanted reverberative sound qualities in a resulting sound signal. Together, techniques envisioned by the present invention produce improved, speed adjusted sound signals.

Methods and devices for selectively generating time-scaled sound signals

A receiver for receiving wireless packetized audio signals having packet headers, and packet payloads including encoded audio signals is provided. The receiver includes an audio decoder for decoding the encoded audio signals into digital audio signals, an audio error detector for detecting impulsive noise in the digital audio signals caused by corrupted payload packets, and a digital signal restoration unit for removing the impulsive noise from the digital audio signals. Further, a method of noise reduction includes the steps of receiving radio frequency packetized audio signals having packet headers, and packet payloads including encoded audio signals and demodulating the RF signals. The method further includes decoding the encoded audio signals into digital audio signals and detecting impulsive noise in the digital audio signals caused by corrupted payload packets. The method further includes removing the impulsive noise by reconstructing the corrupted digital audio signal.

System and method of noise reduction in receiving wireless transmission of packetized audio signals

Methods and systems for a mixed time-frequency domain compander, where the expander and compressor are applied in the frequency domain (subbands), while the limiter and the linear gain are applied in the time-domain (full band). Far-end noise reduction is achieved by adapting the expander gain to the far-end noise, near-end noise compensation is achieved by adapting the linear gain and the compressor gain to the near-end noise. The range of the linear gain section is set individually for each subband, but the linear gain is essentially applied to the full band signal via a scaling procedure. The scaling procedure first reduces the compander gain in the frequency domain by a scaling factor and then performs the reciprocal reverse scaling in the time domain. Far-end noise reduction is coupled to near-end noise compensation in order to avoid cross-modulation of the far-end noise by the near-end noise.

Systems and methods for far-end noise reduction and near-end noise compensation in a mixed time-frequency domain compander to improve signal quality in communications systems

Estimating the noise components of a signal

An apparatus for enhancing intelligibility of a voice signal in a noisy environment includes a first noise estimator which estimates a far-end noise component from the far-end signal, and a second noise estimator which estimates a near-end noise component from the near-end signal. A noise reduction calculator determines a noise reduction gain from the estimated far-end noise, and an echo gain calculator determines an echo control gain. A master gain calculator combines both the echo control gain and the noise control gain into a master gain which is applied to the far-end signal. A comfort noise generator applies the pre-set minimum threshold of the master gain and constantly matches the spectrum of the far-end noise to synthesize a background noise for selectively mixing onto the far-end signal when echo is determined. Echo is determined based on a comparison of the near-end signal and the far-end signal spectra and by compensating the total gain applied. An automatic level controller adjusts the far-end signal responsive to the far-end signal level, and a noise compensator adjusts the far-end signal level responsive to the near-end noise, both adjustments being responsive to the identification of the far-end signal as either speech or noise.

Method and apparatus for network speech enhancement

Signal reconstruction methods reconstruct a missing portion of a signal from a first known portion of the signal preceding the missing portion, and a second known portion of the signal succeeding the missing portion. The first known portion of the signal is represented using a first autoregressive model and the second known portion of the signal is represented using a second autoregressive model. These methods construct a signal estimate from the first and second autoregressive models, thereby permitting the reconstruction of any signal having a missing portion, irrespective of whether or not the length of the missing portion is greater than the short-term stationarity duration of the signal.

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Wayside, NJ, United States of America

Walter Etter