Abstract: A system includes a memory configured to store data associated with a service that is available. The system also includes a microphone associated with an acoustic space and configured to receive an audio input produced by a person. The system further includes a sensor located within the acoustic space and configured to detect vibrations produced by the person. The system includes a processor coupled to the memory, to the microphone, and to the sensor. The processor is configured to conditionally authorize execution of the service requested by the person, the service conditionally authorized based on the audio input and the vibrations.

Abstract: A multichannel acoustic system (MAS) comprises an arrangement of microphones and loudspeakers and a multichannel acoustic processor (MAP) to together enhance conversational speech between two or more persons in a shared acoustic space such as an automobile. The enhancements are achieved by receiving sound signals substantially originating from relatively near sound sources; filtering the sound signals to cancel at least one echo signal detected for at least one microphone from among the plurality of microphones; filtering the sound signals received by the plurality of microphones to cancel at least one feedback signal detected for at least one microphone from among the plurality of microphones; and reproducing the filtered sound signals for each microphone from among the plurality of microphones on a subset of loudspeakers corresponding that are relatively far from the source microphone.

Abstract: Embodiments of an ultrasonic button and methods for using the ultrasonic button are disclosed. In one embodiment, an ultrasonic button may include an ultrasonic transmitter configured to transmit an ultrasonic wave, a piezoelectric receiver layer configured to receive a reflected wave of the ultrasonic wave, a platen layer configured to protect the ultrasonic transmitter and the piezoelectric receiver layer, a first matching layer configured to match an acoustic impedance of the platen layer with an acoustic impedance of ridges of a finger, and an ultrasonic sensor array configured to detect the finger using the reflected wave.

Abstract: A multichannel acoustic system (MAS) comprises an arrangement of microphones, loudspeakers, and filters along with a multichannel acoustic processor (MAP) and other components to together provide and enhance the auditory experience of persons in a shared acoustic space such as, for example, the driver and other passengers in an automobile. Driver-specific features such as navigation and auditory feedback cues are described, as individual auditory customizations and collective communications both within the shared acoustic space as well as with other individuals not located in the space via enhanced conference call facilities.

Abstract: This disclosure provides systems, methods and apparatus related to touch and gesture recognition with an electronic interactive display. The interactive display has a front surface that includes a viewing area, a planar light guide disposed proximate to and behind the front surface, a light source, and at least one photo sensing element coupled with the first planar light guide. The planar light guide is configured to receive scattered light, the received scattered light resulting from interaction between light emitted by the light source and an object in optical contact with the front surface. The photo sensing element is configured to detect at least some of the received scattered light and to output, to a processor, image data. The processor is configured to recognize, from the image data, one or both of a contact pressure and a rotational orientation of the object.

Abstract: A system includes a memory configured to store data associated with a service that is available. The system also includes a microphone associated with an acoustic space and configured to receive an audio input produced by a person. The system further includes a sensor located within the acoustic space and configured to detect vibrations produced by the person. The system includes a processor coupled to the memory, to the microphone, and to the sensor. The processor is configured to conditionally authorize execution of the service requested by the person, the service conditionally authorized based on the audio input and the vibrations.

Abstract: This disclosure provides systems, methods and apparatus related to touch and gesture recognition with an electronic interactive display. The interactive display has a front surface that includes a viewing area, a planar light guide disposed proximate to and behind the front surface, a light source, and at least one photo sensing element coupled with the first planar light guide. The planar light guide is configured to receive scattered light, the received scattered light resulting from interaction between light emitted by the light source and an object in optical contact with the front surface. The photo sensing element is configured to detect at least some of the received scattered light and to output, to a processor, image data. The processor is configured to recognize, from the image data, one or both of a contact pressure and a rotational orientation of the object.

Abstract: Embodiments of an ultrasonic button and methods for using the ultrasonic button are disclosed. In one embodiment, an ultrasonic button may include an ultrasonic transmitter configured to transmit an ultrasonic wave, a piezoelectric receiver layer configured to receive a reflected wave of the ultrasonic wave, a platen layer configured to protect the ultrasonic transmitter and the piezoelectric receiver layer, a first matching layer configured to match an acoustic impedance of the platen layer with an acoustic impedance of ridges of a finger, and an ultrasonic sensor array configured to detect the finger using the reflected wave.

Abstract: A method and system for resynchronizing an embedded multimedia system using bytes consumed in an audio decoder. The bytes consumed provides a mechanism to compensate for bit error handling and correction in a system that does not require re-transmission. The audio decoder keeps track of the bytes consumed and periodically reports the bytes consumed. A host microprocessor indexes the actual bytes consumed since bit errors may have been handled or corrected to a predetermined byte count to determine whether resynchronization is necessary.

Abstract: Voice activity detection using multiple microphones can be based on a relationship between an energy at each of a speech reference microphone and a noise reference microphone. The energy output from each of the speech reference microphone and the noise reference microphone can be determined. A speech to noise energy ratio can be determined and compared to a predetermined voice activity threshold. In another embodiment, the absolute value of the autocorrelation of the speech and noise reference signals are determined and a ratio based on autocorrelation values is determined. Ratios that exceed the predetermined threshold can indicate the presence of a voice signal. The speech and noise energies or autocorrelations can be determined using a weighted average or over a discrete frame size.

Abstract: Power savings in a mobile device is accomplished by generating audio samples by decoding a bitstream with a decoding system within the mobile device. The generated audio samples are transferred into at least one memory bank in a set of memory banks in a power saver block within the mobile device. Parts of the decoding system not involved in the storing of the generated audio samples are switched off after batch decoding a bitstream associated with multiple audio frames. The bitstream includes bits less than that found in one audio file. At least one of the memory banks in the set of memory banks is power collapsible. The fetching of the decoded by the decoding system can be synchronized with a paging channel of a modem in the mobile device. The transferred audio samples is a lossless compression and may occur after a re-encoding.

Abstract: In accordance with a method for providing a distinct perceptual location for an audio source within an audio mixture, a foreground signal may be processed to provide a foreground perceptual angle for the foreground signal. The foreground signal may also be processed to provide a desired attenuation level for the foreground signal. A background signal may be processed to provide a background perceptual angle for the background signal. The background signal may also be processed to provide a desired attenuation level for the background signal. The foreground signal and the background signal may be combined into an output audio source.

Abstract: In a digital system with more than one clock source, lack of synchronization between the clock sources may cause overflow or underflow in sample buffers, also called sample slipping. Sample slipping may lead to undesirable artifacts in the processed signal due to discontinuities introduced by the addition or removal of extra samples. To smooth out discontinuities caused by sample slipping, samples are filtered to when a buffer overflow condition occurs, and the samples are interpolated to produce additional samples when a buffer underflow condition occurs. The interpolated samples may also be filtered. The filtering and interpolation operations can be readily implemented without adding significant burden to the computational complexity of a real-time digital system.

Abstract: Encoder-assisted frame loss concealment (FLC) techniques for decoding audio signals are described. A decoder may discard an erroneous frame of an audio signal and may implement the encoder-assisted FLC techniques in order to accurately conceal the discarded frame based on neighboring frames and side-information transmitted from the encoder. The encoder-assisted FLC techniques include estimating magnitudes of frequency-domain data for the frame based on frequency-domain data of neighboring frames, and estimating signs of the frequency-domain data based on a subset of signs transmitted from the encoder as side-information. Frequency-domain data for a frame of an audio signal includes tonal components and noise components. Signs estimated from a random signal may be substantially accurate for the noise components of the frequency-domain data.

Abstract: An executable is downloaded to an audio output device over a communications link. The executable may configure the audio output device to decode audio encoded in a specified format. The executable may also or alternatively include other audio processing software. The audio may include voice and/or audio playback, e.g., music playback. The ability to download an audio executable allows dynamic provisioning of various decoding and/or audio process capabilities to an audio output device. This may eliminate the need to transcode digitized audio for playback at the audio output device, and may also allow the audio output device to decode multiple audio formats without having multiple audio decoders permanently residing within the audio output device.

Abstract: A method for providing an interface to a processing engine that utilizes intelligent audio mixing techniques may include receiving a request to change a perceptual location of an audio source within an audio mixture from a current perceptual location relative to a listener to a new perceptual location relative to the listener. The audio mixture may include at least two audio sources. The method may also include generating one or more control signals that are configured to cause the processing engine to change the perceptual location of the audio source from the current perceptual location to the new perceptual location via separate foreground processing and background processing. The method may also include providing the one or more control signals to the processing engine.

Abstract: A unified filter bank for performing signal conversions may include an interface that receives signal conversion commands in relation to multiple types of compressed audio bitstreams. The unified filter bank may also include a reconfigurable transform component that performs a transform as part of signal conversion for the multiple types of compressed audio bitstreams. The unified filter bank may also include complementary modules that perform complementary processing as part of the signal conversion for the multiple types of compressed audio bitstreams. The unified filter bank may also include an interface command controller that controls the configuration of the reconfigurable transform component and the complementary modules.

Abstract: Multiple microphone noise suppression apparatus and methods are described herein. The apparatus and methods implement a variety of noise suppression techniques and apparatus that can be selectively applied to signals received using multiple microphones. The microphone signals received at each of the multiple microphones can be independently processed to cancel echo signal components that can be generated from a local audio source. The echo cancelled signals may be processed by some or all modules within a signal separator that operates to separate or otherwise isolate a speech signal from noise signals. The signal separator can include a pre-processing de-correlator followed by a blind source separator. The output of the blind source separator can be post filtered to provide post separation de-correlation. The separated speech and noise signals can be non-linearly processed for further noise reduction, and additional post processing can be implemented following the non-linear processing.

Abstract: This disclosure describes audio mixing techniques that intelligently combine two or more audio signals into an output signal. The techniques allow audio to be combined, yet create perceptual differentiation between the different audio signals. The result is that a user is able to hear both audio signals in a combined output, but the different audio signals do not perceptually interfere with one another. The techniques are relatively simple to implement and are well suited for radio telephones.

Abstract: This disclosure describes signal processing techniques that can improve the performance of blind source separation (BSS) techniques. In particular, the described techniques propose pre-processing steps that can help to de-correlate the different signals from one another prior to execution of the BSS techniques. In addition, the described techniques also propose optional post-processing steps that can further de-correlate the different signals following execution of the BSS techniques. The techniques may be particularly useful for improving BSS performance with highly correlated audio signals, e.g., from two microphones that are in close spatial proximity to one another.