Abstract: Digital signal processing for microphone partial occlusion detection is described. In one embodiment, an electronic system for audio noise processing and for noise reduction, using a plurality of microphones, includes a first noise estimator to process a first audio signal from a first one of the microphones, and generate a first noise estimate. The electronic system also includes a second noise estimator to process the first audio signal, and a second audio signal from a second one of the microphones, in parallel with the first noise estimator, and generate a second noise estimate. A microphone partial occlusion detector determines a low frequency band separation of the first and second audio signals and a high frequency band separation of the first and second audio signals to generate a microphone partial occlusion function that indicates whether one of the microphones is partially occluded.

Abstract: Electronic system for audio noise processing and noise reduction comprises: first and second noise estimators, selector and attenuator. First noise estimator processes first audio signal from voice beamformer (VB) and generate first noise estimate. VB generates first audio signal by beamforming audio signals from first and second audio pick-up channels. Second noise estimator processes first and second audio signal from noise beamformer (NB), in parallel with first noise estimator and generates second noise estimate. NB generates second audio signal by beamforming audio signals from first and second audio pick-up channels. First and second audio signals include frequencies in first and second frequency regions. Selector's output noise estimate may be a) second noise estimate in the first frequency region, and b) first noise estimate in the second frequency region. Attenuator attenuates first audio signal in accordance with output noise estimate. Other embodiments are also described.

Abstract: A method for adapting a threshold used in multi-channel audio voice activity detection. Strengths of primary and secondary sound pick up channels are computed. A separation, being a measure of difference between the strengths of the primary and secondary channels, is also computed. An analysis of the peaks in separation is performed, e.g. using a leaky peak capture function that captures a peak in the separation and then decays over time, or using a sliding window min-max detector. A threshold that is to be used in a voice activity detection (VAD) process is adjusted, in accordance with the analysis of the peaks. Other embodiments are also described and claimed.

Abstract: Methods and apparatuses for comparing a level of a signal carrying ambient acoustic noise with a threshold level; and based on results of the comparison, ignoring time intervals identified as noise burst in estimating ambient noise levels.

Abstract: Methods and apparatuses for deriving a signal-to-noise ratio based at least in part on a measured level of a signal carrying far-end speech, and a measured level of a signal carrying ambient acoustic noise; determining a target gain adjustment based at least in part on the derived signal-to-noise ratio; applying the target gain adjustment to the signal carrying far-end speech to produce a gain-adjusted signal; and providing the gain-adjusted signal for audio output from a communications device.

Abstract: A system for combining signals includes a first microphone generating a first input signal having a first voice component and a first noise component, a second microphone generating a second input signal having a second voice component and a second noise component, a mixing circuit, and an adaptive filter. The mixing circuit applies a first gain having a value ? to the first input signal to produce a first scaled signal, applies a second gain having a value 1?? to the second input signal to produce a second scaled signal, and sums the first scaled signal and the second scaled signal to produce a summed signal. The adaptive filter computes an updated value of ? to minimize the energy of the summed signal based on the summed signal, the first input signal and the second input signal, and provides the updated value of ? to the mixing circuit.

Abstract: Digital signal processing techniques for automatically reducing audible noise from a sound recording that contains speech. A noise suppression system uses two types of noise estimators, including a more aggressive one and less aggressive one. Decisions are made on how to select or combine their outputs into a usable noise estimate in a different speech and noise conditions. A 2-channel noise estimator is described. Other embodiments are also described and claimed.

Abstract: A system for combining signals includes a first microphone generating a first input signal having a first voice component and a first noise component, a second microphone generating a second input signal having a second voice component and a second noise component, a mixing circuit, and an adaptive filter. The mixing circuit applies a first gain having a value ? to the first input signal to produce a first scaled signal, applies a second gain having a value 1?? to the second input signal to produce a second scaled signal, and sums the first scaled signal and the second scaled signal to produce a summed signal. The adaptive filter computes an updated value of ? to minimize the energy of the summed signal based on the summed signal, the first input signal and the second input signal, and provides the updated value of ? to the mixing circuit.

Abstract: Methods and apparatuses for deriving a signal-to-noise ratio based at least in part on a measured level of a signal carrying far-end speech, and a measured level of a signal carrying ambient acoustic noise, determining a target gain adjustment based at least in part on the derived signal-to-noise ratio, including determining a coarse gain adjustment and a fine gain adjustment, the target gain adjustment corresponding to a combination of the coarse gain adjustment and the fine gain adjustment, applying the coarse gain adjustment to the signal carrying far-end speech using first gain adjustment circuitry to produce a first gain-adjusted signal, applying the fine gain adjustment to the signal carrying far-end speech using second gain adjustment circuitry to produce a second gain-adjusted signal, and providing a result of combining the first gain-adjusted signal and the second gain-adjusted signal to audio output from a wireless communications device.

Abstract: Methods and apparatuses for deriving a signal-to-noise ratio based at least in part on a measured level of a signal carrying far-end speech, and a measured level of a signal carrying ambient acoustic noise; determining a target gain adjustment based at least in part on the derived signal-to-noise ratio; applying the target gain adjustment to the signal carrying far-end speech to produce a gain-adjusted signal; and providing the gain-adjusted signal for audio output from a communications device.

Abstract: Methods and apparatuses for comparing a level of a signal carrying ambient acoustic noise with a threshold level; and based on results of the comparison, ignoring time intervals identified as noise burst in estimating ambient noise levels.

Abstract: A microphone includes a sensing element having two opposing sides; and a housing including a first acoustic port having an external-facing portion defined in part by a first aperture located on a first housing side and an internal-facing portion defined in part by a first cavity within the housing, the first cavity being coupled to a first side of the element; and a second acoustic port having an external-facing portion defined in part by a second aperture located on the first housing side and an internal-facing portion defined in part by a second cavity within the housing, the second cavity being coupled to a second side of the element. The ports are spaced apart at a distance such that a level of an electrical response by the element to an ambient acoustic noise at 50 dB A-weighted sound pressure level exceeds an internal electrical noise level of the element.

Abstract: An automotive system provides an integrated user interface for control and communication functions in an automobile or other type of vehicle. The user interface supports voice enabled interactions, as well as other modes of interaction, such as manual interactions using controls such as dashboard or steering wheel mounted controls. The system also includes interfaces to devices in the vehicle, such as wireless interfaces to mobile devices that are brought into the vehicle. The system also provides interfaces to information sources such as a remote server, for example, for accessing information.

Abstract: A digital telephone answering device having speakerphone capability allows a recorded message to be played back and heard by the far-end party as well as over the local speakerphone by the near-end party during speakerphone operation as if it were a normal receive signal. Moreover, normal speakerphone conversation is possible during this conversational playback mode allowing the far-end party and/or near-end party to break-in over the played back pre-recorded message as desired and be heard by the other party. Both message and conversation signals are preferably (but not necessarily) at similar levels. Also, the message playback at the near end is preferably (but not necessarily) subject to the same speakerphone digital volume control as that received at the far end.

Abstract: A digital telephone answering device having speakerphone capability allows a recorded message to be played back and heard by the far-end party as well as over the local speakerphone by the near-end party during speakerphone operation as if it were a normal receive signal. Moreover, normal speakerphone conversation is possible during this conversational playback mode allowing the far-end party and/or near-end party to break-in over the played back pre-recorded message as desired and be heard by the other party. Both message and conversation signals are preferably (but not necessarily) at similar levels. Also, the message playback at the near end is preferably (but not necessarily) subject to the same speakerphone digital volume control as that received at the far end.

Abstract: An automotive system provides an integrated user interface for control and communication functions in an automobile or other type of vehicle. The user interface supports voice enabled interactions, as well as other modes of interaction, such as manual interactions using controls such as dashboard or steering wheel mounted controls. The system also includes interfaces to devices in the vehicle, such as wireless interfaces to mobile devices that are brought into the vehicle. The system also provides interfaces to information sources such as a remote server, for example, for accessing information.

Abstract: A configurable adaptive echo canceler which can perform acoustical echo cancellation (AEC) or hybrid echo cancellation (HEC) under software control. Particular parameters based on the particular application can be associated with either a suitable AEC or a suitable HEC, and the parameters are made selectable through the use of a control register. Exemplary parameters are accuracy of filter coefficients, a span select to manipulate the expected length of the echo path, a step size to adjust the adaptation of the echo cancellation, enablement of a tap update, selection of adaptation mode, center clipper selection, and sub-band or full-band echo cancellation. With the use of audio line switching, the configurable echo canceler can allow various modes of operation of the telephone, some of which may require an AEC in the installed unit and/or some of which may require an HEC.

Abstract: A digital cordless telephone combines speakerphone control information together with voice data information in each of a plurality of data packets transmitted between the remote handset and the base unit while the remote handset is operating in a speakerphone mode. To allow half-duplex speakerphone functionality in the digital remote handset, the speakerphone control information includes break-in information relating to a break in by the user of the remote handset into an ongoing conversation. To allow full-duplex, echo cancellation mode operation of the speakerphone functionality at the digital remote handset, the speakerphone control information includes coordinating information relating to an amount of echo cancellation performed by the base unit and by the remote handset. In the disclosed embodiment, the speakerphone control information further includes switching and signal level information.

Abstract: A telephone answering device including two separate coders, a first coder for encoding/decoding fixed voice prompts spoken by a single speaker, and a second coder for encoding/decoding incoming and outgoing voice messages spoken by multiple speakers. The first coder uses a first set of codebooks trained based on a first set of utterances spoken by a single speaker, while the second coder uses a second set of codebooks trained based on a second set of utterances spoken by multiple speakers. Because the first set of utterances is significantly smaller in size than the second set of utterances, and the range of pitch period is significantly smaller in size for the first set of utterances spoken by a single speaker in comparison to that of the second set of utterances spoken by multiple speakers, the size of the first set of codebooks is significantly reduced relative to the size of the second set of codebooks.

Abstract: In one embodiment according to the invention, a digital cordless telephone includes a base unit adapted to couple to a telephone network, to transmit an outgoing message to an incoming caller, to store a digital message from the incoming caller, and to transmit and receive digital samples over an RF link. The digital cordless telephone of this embodiment also includes a handset adapted to transmit and receive digital samples over the RF link. According to this embodiment, the base unit and handset are further adapted to enable a user to monitor the stored digital message from the handset.