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 mobile device uses externals microphone signals to improve the estimate of background noise that it computes. In order to improve voice quality in a first signal that is produced by an internal microphone, the mobile device identifies an external microphone device within proximity of the mobile device. The mobile device establishes a wireless connection with the external microphone device. The mobile device receives a second signal from the external microphone device through the wireless connection. The second signal is produced by a microphone of the external microphone device. The mobile device generates a noise profile based on the second signal, and then suppresses background/ambient noise from the first signal based on the noise profile. Other embodiments are also described.
Abstract:
Various techniques for adaptively varying audio feedback data on an electronic device are provided. In one embodiment, an audio user interface implementing certain aspects of the present disclosure may devolve or evolve the verbosity of audio feedback in response to user interface events based at least partially upon the verbosity level of audio feedback provided during previous occurrences of the user interface event. In another embodiment, an audio user interface may be configured to vary the verbosity of audio feedback associated with a navigable list of items based at least partially upon the speed at which a user navigates the list. In a further embodiment, an audio user interface may be configured to vary audio feedback verbosity based upon the contextual importance of a user interface event. Electronic devices implementing the present techniques provide an improved user experience with regard to audio user interfaces.
Abstract:
Implementations of the subject technology provide systems and methods for determining whether to interrupt a user of an audio device that is operating in a noise-cancelling mode of operation. For example, the user may desire to be interrupted by one or more pre-designated contacts that are identified at an associated electronic device as interrupt-authorized contacts, or by a person who speaks a designated keyword to the user.
Abstract:
Aspects of the subject technology relate to a device including a microphone, a filter and a processor. The filter receives an audio signal including ambient noise and a voice of a user of the device from the microphone. At least a portion of ambient noise is filtered from the audio signal. The processor determines a level of the ambient noise in the received audio signal and dynamically adjusts a gain applied to the filtered audio signal based on the level of the ambient noise.
Abstract:
Method and system for maintaining accuracy of voice recognition are described herein. The audio system reproducing sound using a loudspeaker array that is housed in a loudspeaker cabinet may selection from a number of sound rendering modes and changing the selected sound rendering mode based on the current playback volume set on the audio system. The sound rendering modes include at least one of: a number of free space modes and a number of complex modes. Other aspects are also described and claimed.
Abstract:
A system and method is described for determining whether a loudspeaker device has relocated, tilted, rotated, or changed environment such that one or more parameters for driving the loudspeaker may be modified and/or a complete reconfiguration of the loudspeaker system may be performed. In one embodiment, the system may include a set of sensors. The sensors provide readings that are analyzed to determine 1) whether the loudspeaker has moved since a previous analysis and/or 2) a distance of movement and/or a degree change in orientation of the loudspeaker since the previous analysis. Upon determining the level of movement is below a threshold value, the system adjusts previous parameters used to drive one or more of the loudspeakers. By adjusting previous parameters instead of performing a complete recalibration, the system provides a more efficient technique for ensuring that the loudspeakers continue to produce accurate sound for the listener.
Abstract:
An automatic speech recognition (ASR) triggering system, and a method of providing an ASR trigger signal, is described. The ASR triggering system can include a microphone to generate an acoustic signal representing an acoustic vibration and an accelerometer worn in an ear canal of a user to generate a non-acoustic signal representing a bone conduction vibration. A processor of the ASR triggering system can receive an acoustic trigger signal based on the acoustic signal and a non-acoustic trigger signal based on the non-acoustic signal, and combine the trigger signals to gate an ASR trigger signal. For example, the ASR trigger signal may be provided to an ASR server only when the trigger signals are simultaneously asserted. Other embodiments are also described and claimed.
Abstract:
A speech recognition system for resolving impaired utterances can have a speech recognition engine configured to receive a plurality of representations of an utterance and concurrently to determine a plurality of highest-likelihood transcription candidates corresponding to each respective representation of the utterance. The recognition system can also have a selector configured to determine a most-likely accurate transcription from among the transcription candidates. As but one example, the plurality of representations of the utterance can be acquired by a microphone array, and beamforming techniques can generate independent streams of the utterance across various look directions using output from the microphone array.
Abstract:
A method for controlling a speech enhancement process in a far-end device, while engaged in a voice or video telephony communication session over a communication link with a near-end device. A near-end user speech signal is produced, using a microphone to pick up speech of a near-end user, and is analyzed by an automatic speech recognizer (ASR) without being triggered by an ASR trigger phrase or button. The recognized words are compared to a library of phrases to select a matching phrase, where each phrase is associated with a message that represents an audio signal processing operation. The message associated with the matching phrase is sent to the far-end device, which is used to configure the far-end device to adjust the speech enhancement process that produces the far-end speech signal. Other embodiments are also described.