Abstract: A wireless system including a first speaker and a second speaker, where the first and second speakers communicate with each other over a wireless link. In some configurations, the first speaker includes both a primary wireless interface for receiving audio from an audio source and a secondary wireless interface transmitting a portion of the audio to the second speaker. The speaker can incorporate Near Field Communication (NFC) technology to provide the wireless link between each other. The wireless system can be configured to synchronize audio output at the speakers, and can also include a second-speaker detection mechanism that permits the first speaker to be used in either a stand-alone mode, with audio output at only the first speaker, or full-headset mode, with audio output at both speakers when the second speaker is detected within wireless range of the first speaker.

Abstract: A method of initiating outbound calls through a telephony interface. The method accepts an inbound call from a user, authenticates the inbound call against a predetermined user identity, accesses an address book associated with the user identity, receives an spoken request from the user, selects a telephone record in the address book associated with the spoken request, initiates an outbound call to the telephone record; and connects the inbound call with the outbound call responsive to a connection established on the outbound call. In an alternative embodiment the method accepts an inbound call, authenticates the inbound call against a predetermined user identity, receives a dual-tone frequency-modulated input representing a desired outbound call target, initiates an outbound call to the outbound call target, and connects the inbound call with the outbound call responsive to a connection established on the outbound call. A corresponding computer system for each embodiment is also provided.

Abstract: A method for processing a multichannel audio signal may be configured to control the amplitude of one channel of the signal relative to another based on the levels of the two channels. One such example uses a bias factor, which is based on a standard orientation of an audio sensing device relative to a directional acoustic information source, for amplitude control of information segments of the signal.

Abstract: An enhanced blind source separation technique is provided to improve separation of highly correlated signal mixtures. A beamforming algorithm is used to precondition correlated first and second input signals in order to avoid indeterminacy problems typically associated with blind source separation. The beamforming algorithm may apply spatial filters to the first signal and second signal in order to amplify signals from a first direction while attenuating signals from other directions. Such directionality may serve to amplify a desired speech signal in the first signal and attenuate the desired speech signal from the second signal. Blind source separation is then performed on the beamformer output signals to separate the desired speech signal and the ambient noise and reconstruct an estimate of the desired speech signal. To enhance the operation of the beamformer and/or blind source separation, calibration may be performed at one or more stages.

Abstract: Descriptions are provided of various implementations of an automated tuning process configured to optimize a procedure for post-processing images captured by a camera sensor.

Abstract: Techniques of this disclosure provide for adjustment of a conversion rate of a sampling rate converter (SRC) in real-time. The SRC determines relative timing of generated output samples based on non-approximated integer components that are recursively updated. The SRC may further base relative timing of output samples on a value of one or more step size components associated with the integer components. Also according to techniques of this disclosure, a conversion rate of an SRC may be adjusted in real-time based on a detected mismatch between a source clock of a digital input signal and a local clock.

Abstract: Systems, methods, and apparatus for the detection of signals having spectral peaks with narrow bandwidth are described herein. The range of described configurations includes implementations that perform such detection using parameters of a linear prediction coding (LPC) analysis scheme.

Abstract: Estimating the proximity of an audio source is accomplished by transforming audio signals from a plurality of sensors to frequency domain. The amplitudes of the transformed audio signals are then determined. The proximity of the audio source is determined based on a comparison of the frequency domain amplitudes. This estimation permits a device to differentiate between relatively distant audio sources and audio sources at close proximity to the device. The technique can be applied to mobile handsets, such as cellular phones or PDAs, hands-free headsets, and other audio input devices. Devices taking advantage of this “close proximity” detection are better able to suppress background noise and deliver an improved user experience.

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: A method for detecting wind noise is described. At least two audio signals are received. The at least two audio signals are filtered to reduce higher frequencies and to reduce lower frequencies to provide at least two filtered audio signals. The cross correlation of the at least two filtered audio signals is computed for multiple delays. A maximum cross correlation is determined from the cross correlations computed for the multiple delays. Wind noise is detected by comparing the maximum cross correlation with a threshold.

Abstract: Systems, methods, and apparatus for processing an M-channel input signal are described that include outputting a signal produced by a selected one among a plurality of spatial separation filters. Applications to separating an acoustic signal from a noisy environment are described, and configurations that may be implemented on a multi-microphone handheld device are also described.

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: Methods, apparatus, and systems for source separation include a converged plurality of coefficient values that is based on each of a plurality of M-channel signals. Each of the plurality of M-channel signals is based on signals produced by M transducers in response to at least one information source and at least one interference source. In some examples, the converged plurality of coefficient values is used to filter an M-channel signal to produce an information output signal and an interference output signal.

Abstract: A wireless headset includes a first earpiece and a second earpiece, where the first and second earpieces communicate with each other over a wireless link. In some configurations, the first earpiece includes both a primary wireless interface for receiving audio from an audio source and a secondary wireless interface transmitting a portion of the audio to the second earpiece. The earpieces can incorporate Near Field Communication (NFC) technology to provide the wireless link between each other. The wireless headset can be configured to synchronize audio output at the earpieces, and can also include a second-earpiece detection mechanism that permits the first earpiece to be used in either a stand-alone mode, with audio output at only the first earpiece, or full-headset mode, with audio output at both earpieces when the second earpiece is detected within wireless range of the first earpiece.

Abstract: Systems, methods, and apparatus described include waveform alignment operations in which a single set of evaluated cosines and sines is used to calculate cross-correlations of two periodic waveforms at two different phase shifts.

Abstract: Subject matter disclosed herein relates to setting photographic parameters based, at least in part, on signal measurements, such as, for example, without limitation, position of the Sun. Briefly, in accordance with one embodiment, an apparatus may comprise the following: a wireless communication capable device (E.G., 400, 1100) including an integrated digital camera (E.G., 1180, 402); the device capable of estimating position of the Sun relative to the position of the device based at least in part on signal information received via wireless communication; and the device being further capable of using the estimated position of the Sun so as to affect one or more photographic parameters. Briefly, in accordance with another embodiment, a method may comprise the following: acquiring information via wireless communication (E.G.

Abstract: A camera system in normal mode and hand jitter reduction (hjr) mode may comprise generating a first exposure time-gain product by multiplying the normal mode exposure time with the normal mode gain. It may further comprise modifying the normal mode exposure time and gain and multiplying these modified parameters to generate a second exposure time-gain product for a hjr mode that reduces the difference between the first exposure time-gain product and the second exposure time-gain product. To reduce the difference the normal mode frame rate may also be modified. Operation of a camera in normal mode may be in response to a sensed light level being above a threshold. The hjr mode may be selected by the user while the camera is operating. The hjr mode may be used in response to a sensed light level being lower than the threshold.

Abstract: Power consumption may be reduced in a media device including a first processor coupled to the non-volatile memory, either directly or indirectly, allowing the first processor to generate a pointer structure. The first processor may also be coupled, either directly or indirectly to a memory space, allowing the first processor to write the pointer structure in the memory space. The media device includes a second processor, such as a DSP/SHW or peripheral processor, and may be also be coupled, either directly or indirectly to the memory space, allowing the second processor to retrieve a block of media data from the non-volatile memory. Retrieval of the block of media data may be read directly from the non-volatile memory, or in some cases, the media data being retrieved may be parsed. The media data may be an audio file data, video file data, or both.

Abstract: A method for improving the similarity of the volumes in different audio players is described. First player metrics for one or more Musical Instrument Digital Interface (MIDI) instruments may be determined. A digital music file that uses the MIDI protocol may be received. A note parameter or channel parameter may be adjusted for notes in the digital music file based on the first player metrics.

Abstract: This disclosure describes techniques for processing audio files that comply with the musical instrument digital interface (MIDI) format. In particular, this disclosure describes storage of MIDI parameters for efficient access by a processor and a hardware unit. The processor may be a digital signal processor (DSP) and the hardware unit may be specifically designed to process MIDI parameters. In one aspect, this disclosure provides an apparatus comprising a processor that converts a MIDI event into MIDI parameters, a hardware unit that uses MIDI parameters to generate audio samples, and a plurality of storage units that store MIDI parameters which are accessible by both the processor and the hardware unit.