Abstract: Described herein are multi-channel noise suppression systems and methods that are configured to detect and suppress wind and background noise using at least two spatially separated microphones: at least one primary speech microphone and at least one noise reference microphone. The multi-channel noise suppression systems and methods are configured, in at least one example, to first detect and suppress wind noise in the input speech signal picked up by the primary speech microphone and, potentially, the input speech signal picked up by the noise reference microphone. Following wind noise detection and suppression, the multi-channel noise suppression systems and methods are configured to perform further noise suppression in two stages: a first linear processing stage that includes a blocking matrix and an adaptive noise canceler, followed by a second non-linear processing stage.

Abstract: Multi-channel noise suppression systems and methods are described that omit the traditional delay-and-sum fixed beamformer in devices that include a primary speech microphone and at least one noise reference microphone with the desired speech being in the near-field of the device. The multi-channel noise suppression systems and methods use a blocking matrix (BM) to remove desired speech in the input speech signal received by the noise reference microphone to get a “cleaner” background noise component. Then, an adaptive noise canceler (ANC) is used to remove the background noise in the input speech signal received by the primary speech microphone based on the “cleaner” background noise component to achieve noise suppression. The filters implemented by the BM and ANC are derived using closed-form solutions that require calculation of time-varying statistics of complex frequency domain signals in the noise suppression system.

Abstract: The present invention is directed to a wireless telephone having a first microphone and a second microphone and a method for processing audio signal in a wireless telephone having a first microphone and a second microphone. The wireless telephone includes a first microphone, a second microphone, and a signal processor. The first microphone outputs a first audio signal, the first audio signal comprising a voice component and a background noise component. The second microphone outputs a second audio signal. The signal processor increases a ratio of the voice component to the noise component of the first audio signal based on the content of at least one of the first audio signal and the second audio signal to produce a third audio signal.

Abstract: Unlike sound based pressure waves that go everywhere, air turbulence caused by wind is usually a fairly local event. Therefore, in a system that utilizes two or more spatially separated microphones to pick up sound signals (e.g., speech), wind noise picked up by one of the microphones often will not be picked up (or at least not to the same extent) by the other microphone(s). Embodiments of methods and apparatuses that utilize this fact and others to effectively detect and suppress wind noise using multiple microphones that are spatially separated are described.

Abstract: Methods, systems, and apparatuses are described for improved multi-microphone source tracking and noise suppression. In multi-microphone devices and systems, frequency domain acoustic echo cancellation is performed on each microphone input, and microphone levels and sensitivity are normalized. Methods, systems, and apparatuses are also described for improved acoustic scene analysis and source tracking using steered null error transforms, on-line adaptive acoustic scene modeling, and speaker-dependent information. Switched super-directive beamforming reinforces desired audio sources and closed-form blocking matrices suppress desired audio sources based on spatial information derived from microphone pairings. Underlying statistics are tracked and used to updated filters and models. Automatic detection of single-user and multi-user scenarios, and single-channel suppression using spatial information, non-spatial information, and residual echo are also described.

Abstract: Methods, systems, and apparatuses are described for performing speaker-identification-assisted speech processing in an uplink path of a communication device. In accordance with certain embodiments, a communication device includes speaker identification (SID) logic that is configured to identify the identity of a near-end speaker. Knowledge of the identity of the near-end speaker is then used to improve the performance of one or more uplink speech processing algorithms implemented on the communication device.

Abstract: Methods, systems, and apparatuses are described for performing speaker-identification-assisted speech processing. In accordance with certain embodiments, a communication device includes speaker identification (SID) logic that is configured to identify a user of the communication device and/or the identity of a far-end speaker participating in a voice call with a user of the communication device. Knowledge of the identity of the user and/or far-end speaker is then used to improve the performance of one or more speech processing algorithms implemented on the communication device.

Abstract: Methods, systems, and apparatuses are described for performing speaker-identification-assisted speech processing in a downlink path of a communication device. In accordance with certain embodiments, a communication device includes speaker identification (SID) logic that is configured to identify the identity of a far-end speaker participating in a voice call with a user of the communication device. Knowledge of the identity of the far-end speaker is then used to improve the performance of one or more downlink speech processing algorithms implemented on the communication device.

Abstract: A system and method for modifying the size of data packets transmitted over a packet network in a manner that avoids overloading of the network. The system and method involves monitoring one or more parameters indicative of an amount of bandwidth being utilized on the packet network, responsive to the monitoring, determining that a level of bandwidth utilization on the packet network has changed, responsive to the determination that the level of bandwidth utilization on the packet network has changed, issuing a command to change the size of packets used for carrying data from a first packet size to a second packet size.

Abstract: Systems and methods are described that apply a noise feedback coding (NFC) technique at the encoder of a delta modulation codec, such as a Continuously Variable Slope Delta Modulation (CVSD) codec, so as to shape the spectrum of the coding noise produced thereby in such a way that the speech quality of the delta modulation decoder output is enhanced. The techniques described herein are not limited to delta modulation codecs and may also be applied to any sample-by-sample codec, including a G.711 ?-law codec, a linear pulse code modulation (LPCM), or any other of a wide variety of sample-by-sample codecs, to improve the audio quality of the decoder output thereof.

Abstract: A system and method is described that improves the intelligibility of a far-end telephone speech signal to a user of a telephony device in the presence of near-end background noise. As described herein, the system and method improves the intelligibility of the far-end telephone speech signal in a manner that does not require user input and that minimizes the distortion of the far-end telephone speech signal. The system is integrated with an acoustic echo canceller and shares information therewith.

Abstract: Systems and methods for enhancing the quality of an audio signal produced by an audio codec are described herein. In accordance with the systems and methods, a pitch-based pre-filter adaptively filters an input audio signal to produce a filtered audio signal. An audio encoder encodes the filtered audio signal to generate a compressed audio bit stream. An audio decoder decodes the compressed audio bit stream to generate a decoded audio signal. A pitch-based post-filter adaptively filters the decoded audio signal to produce an output audio signal, wherein adaptively filtering the decoded audio signal comprises undoing at least part of a signal-shaping effect of the pitch-based pre-filter.

Abstract: A method for overlap-adding signals useful for performing frame loss concealment (FLC) in an audio decoder as well as in other applications. The method uses a dynamic mix of windows to overlap two signals whose normalized cross-correlation may vary from zero to one. If the overlapping signals are decomposed into a correlated component and an uncorrelated component, they are overlap-added separately using the appropriate window, and then added together. If the overlapping signals are not decomposed, a weighted mix of windows is used. The mix is determined by a measure estimating the amount of cross-correlation between overlapping signals, or the relative amount of correlated to uncorrelated signals.

Abstract: Systems and methods are described that utilize dynamic time scale modification (TSM) to achieve reduced bit rate audio coding. In accordance with embodiments, different levels of TSM compression are selectively applied to segments of an input speech signal prior to encoding thereof by an encoder. Encoded TSM-compressed segments are received at a decoder which decodes such segments and then applies an appropriate level of TSM decompression to each based on information received from the encoder. By selectively applying different levels of TSM compression to segments of an input speech signal prior to encoding, a coding bit rate associated with the encoder/decoder is reduced. Furthermore, by selecting a level of TSM compression for each segment of the input speech signal that takes into account certain local characteristics of that signal, such bit rate reduction is provided without introducing unacceptable levels of distortion into an output speech signal produced by the decoder.

Abstract: A system and method is described that improves the intelligibility of a far-end telephone speech signal to a user of a telephony device in the presence of near-end background noise. As described herein, the system and method improves the intelligibility of the far-end telephone speech signal in a manner that does not require user input and that minimizes the distortion of the far-end telephone speech signal. The system is integrated with an acoustic echo canceller and shares information therewith.

Abstract: A method of concealing bit errors in a signal is provided. The method includes decoding an encoded signal parameter based upon constraints placed on a signal parameter, comparing the decoded signal parameter against the constraints, and declaring the decoded signal parameter invalid when the constraints are violated.

Abstract: Systems and methods are described that apply a noise feedback coding (NFC) technique at the encoder of a delta modulation codec, such as a Continuously Variable Slope Delta Modulation (CVSD) codec, so as to shape the spectrum of the coding noise produced thereby in such a way that the speech quality of the delta modulation decoder output is enhanced. The techniques described herein are not limited to delta modulation codecs and may also be applied to any sample-by-sample codec, including a G.711 ?-law codec, a linear pulse code modulation (LPCM), or any other of a wide variety of sample-by-sample codecs, to improve the audio quality of the decoder output thereof.

Abstract: A technique is provided for automatically adjusting the volume, or magnitude, of an audio signal. The technique includes calculating an average power associated with a segment of an input audio signal, determining whether the average power is greater than an estimated signal level associated with one or more previously-processed segments of the input audio signal and, depending on the determination, either calculating an updated estimated signal level by subtracting from the average power an attenuated difference between the estimated signal level and the average power or setting the updated estimated signal level to the average power. A gain to be applied to the segment of the input audio signal is then determined based on the updated estimated signal level and a target signal level for an output audio signal.

Abstract: The present invention is directed to a multiple description transmission system that provides redundancy to combat transmission channel impairments. The multiple description transmission system includes a first and second wireless telephone. The first wireless telephone includes the following: an array of microphones, wherein each microphone in the array of microphones is configured to receive voice input from a user and to produce a voice signal corresponding thereto; an encoder coupled to the microphone array and configured to encode each of the voice signals; and a transmitter coupled to the encoder and configured to transmit each of the encoded voice signals. The second wireless telephone includes the following: a receiver configured to receive the transmitted signals; a decoder coupled to the receiver and configured to decode the signals received by the receiver, thereby producing an output signal; and a loudspeaker that receives the output signal and produces a pressure sound wave corresponding thereto.

Abstract: Systems and methods are described for enhancing the audio quality of an FM receiver. In embodiments described herein, quadrature L?R demodulation is applied to a composite baseband signal output by an FM demodulator to obtain an L?R noise signal. A channel quality measure is calculated based on the L?R noise signal and is used to control whether a pop suppression technique is applied to an L+R signal obtained from the composite baseband signal to detect and remove noise pulses therefrom. The channel quality measure and the L?R noise signal are also leveraged to perform single-channel noise suppression in the frequency domain on an L?R signal obtained from the composite baseband signal and on the L+R signal. The channel quality measure is also used to control the application of a fast fading compensation process that replaces noisy segments of the L?R and L+R signal with replacement waveforms generated via waveform extrapolation.