Abstract: A system, apparatus, method and computer program product determine the location of a receiver, such as one or more sensors, carried by a device, e.g., a robot. In a method, an audio signal is received in response to a predetermined audio signal provided by at least two audio source devices. For the audio signal that is received from a respective audio source device, the method estimates a first impulse response between the respective audio source device and the receiver. For the first impulse response estimated between each respective audio source device of the at least two audio source devices and the receiver, the method removes one or more reflections from the first impulse response to create direct path information. The method also includes determining a location of the receiver based at least in part upon the direct path information.

Abstract: A system, apparatus, method and computer program product determine the location of a receiver, such as one or more sensors, carried by a device, e.g., a robot. In a method, an audio signal is received in response to a predetermined audio signal provided by at least two audio source devices. For the audio signal that is received from a respective audio source device, the method estimates a first impulse response between the respective audio source device and the receiver. For the first impulse response estimated between each respective audio source device of the at least two audio source devices and the receiver, the method removes one or more reflections from the first impulse response to create direct path information. The method also includes determining a location of the receiver based at least in part upon the direct path information.

Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: An automatic gain control derives the gain from the distance between the sound source and the microphone. The distance-based automatic gain control normalizes signal level changes caused by a speaker not maintaining a constant distance with respect to the microphone. Also, a proximity-effect compensation that derives the adaptive filter from the distance between the sound source and the microphone. The proximity-effect compensation corrects frequency response changes due to undesired proximity-effect variations. Determination of the distance between a sound source and a microphone permits accurate compensation for both frequency response changes and distance-related signal level changes.

Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: 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 and in the viewing of videos with associated audio (e.g., movies). In one embodiment, a viewer's location and head position relative to a video display screen is determined, one or more desired sound source locations (which may, for example, be related to a projection on the video display) are determined, and binaural stereo audio signals which accurately locate the sound sources at the desired sound source locations are advantageously generated.

Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: 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.