Abstract: Embodiments may be a standalone module or part of mobile devices, desktop computers, servers, stereo systems, or any other systems that might benefit from condensed audio presentations of item structures such as lists or tables. Embodiments may comprise logic such as hardware and/or code to adjust the temporal characteristics of items comprising words. The items maybe included in a structure such as a text listing or table, an audio listing or table, or a combination thereof, or may be individual words or phrases. For instance, embodiments may comprise a keyword extractor to extract keywords from the items and an abbreviations generator to generate abbreviations based upon the keywords. Further embodiments may comprise a text-to-speech generator to generate audible items based upon the abbreviations to render to a user while traversing the item structure.

Abstract: A mobile user interface with energy harvesting is presented. In one embodiment, the mobile user interface comprises a striker, a piezoelectronic element to generate electric energy in response to being struck by the striker under control of an elastic mechanism, and a transmitter coupled to use the electric energy to transmit a signal wirelessly.

Abstract: A system to generate an enhanced acoustic transmission signal includes a carrier signal generator to generate a carrier signal. A data signal generator is provided to receive data and to generate a data signal representing the data. A signal modulator is also provided to modulate the carrier signal with the data signal to form a modulated carrier signal at a carrier frequency. The system includes a masking signal generator to generate a masking signal to mask the modulated carrier signal from being audible by a human ear. An audio input device is provided to receive audio and to generate an audio signal based on the audio, wherein a frequency band surrounding the carrier frequency is removed from the audio signal. A signal adder is also provided to combine the modulated carrier signal, the masking signal, and the audio signal to form the enhanced acoustic transmission signal.

Abstract: Embodiments for an apparatus and method are provided that can build latency in streaming applications that use data packets. In an embodiment, a system has an under-run forecasting mechanism, a statistics monitoring mechanism, and a playback queuing mechanism. The under-run forecasting mechanism determines an estimate of when a supply of data packets to convert will be exhausted. The statistics monitoring mechanism measures the arrival fluctuations of the supply of data packets. The playback queuing mechanism can build the latency.

Abstract: Embodiments for an apparatus and method are provided for building optimal latency in streaming applications that use data packets. In an embodiment, a system has an under-run forecasting mechanism, a statistics monitoring mechanism, and a playback queuing mechanism. The under-run forecasting mechanism determines an estimate of when a supply of data packets to convert will be exhausted. The statistics monitoring mechanism measures the arrival fluctuations of the supply of data packets. The playback queuing mechanism builds latency in the supply of data packets based on the estimate determined by the under-run forecasting mechanism and the arrival fluctuations measured by the statistics monitoring mechanism. In an embodiment, the supply of data packets relates to audio.

Abstract: An indoor location awareness method for locating a device using device-observable signals of known proximity sensors and device-unobservable signals of known proximity sensors. The exclusion of the possibility that the device is within an unobservable proximity sensor's range increases the knowledge of the device's true location. In addition to defining the location of the device, the device can predict the arrival and departure of unobservable proximity sensors as it moves throughout the environment.

Abstract: An input audio signal is analyzed to determine a power spectral density profile and the power spectral density profile is compared with at least one template profile. On the basis of the comparison, frequency bands of the input audio signal are selectively attenuated.

Abstract: A method and apparatus to perform speech recognition over a voice channel is described.

Abstract: Briefly, in accordance with one embodiment of the invention, a method for actively characterizing the latency of an audio channel of a computer, such as a personal computer, is provided. At least two signal streams for a waveform are created in the audio channel. The presence of the first signal sample stream for the waveform and the second signal sample stream for the waveform is detected at a point in the audio channel. The time between the detections of the signal sample streams is measured. Briefly, in accordance with another embodiment of the invention, a method of actively characterizing the latency of an audio channel of a computer, such as a personal computer, is provided. At least a first and a second waveform are created in the audio channel. The presence of the first and the second waveform are detected at a point in the audio channel. The time between the detections of the waveforms is measured.

Abstract: An echo cancellation unit includes a model store to store a current echo model from an adaptive filter when a real-time error occurs. The real-time error is detected by monitoring a convergence metric. In some embodiments, the convergence metric is echo return loss enhancement (ERLE). When a real-time error occurs, the current echo model is saved, and the adaptive filter is reset such that it will begin converging from the origin. As a new model emerges in the adaptive filter, it is compared to the saved model in the model store at several time lags. If a match is found, the emerging model is replaced with the saved model at the appropriate time lag. The result is faster convergence of the adaptive filter than if the adaptive filter were left to converge on its own.

Abstract: An input audio signal is analyzed to determine a power spectral density profile and the power spectral density profile is compared with at least one template profile. On the basis of the comparison, frequency bands of the input audio signal are selectively attenuated.

Abstract: In an echo cancellation system, an echo canceler has an echo channel signal input, a reference channel signal input and a control output, the two inputs being uncorrelated, and a delay buffer has an input for a reference channel signal, an output in communication with reference channel signal input and a control input in communication with the control output from the echo canceler.

Abstract: Embodiments for an apparatus and method are provided for building optimal latency in streaming applications that use data packets. In an embodiment, a system has an under-run forecasting mechanism, a statistics monitoring mechanism, and a playback queuing mechanism. The under-run forecasting mechanism determines an estimate of when a supply of data packets to convert will be exhausted. The statistics monitoring mechanism measures the arrival fluctuations of the supply of data packets. The playback queuing mechanism builds latency in the supply of data packets based on the estimate determined by the under-run forecasting mechanism and the arrival fluctuations measured by the statistics monitoring mechanism. In an embodiment, the supply of data packets relates to audio.

Abstract: A method of transmitting an audio analog signal over limited bandwidth transmit channels include digitizing the analog signal and splitting the digitized signal into multiple channels. One channel is downsampled and anti-aliased. Another channel is high-pass filtered and downsampled. The channels are then encoded, and multiplexed together along with an alignment signal. The channels are then transmitted over limited bandwidth channels.

Abstract: Building optimal latency in streaming applications that use data packets. In one embodiment, a system having an under-run forecasting mechanism, a statistics monitoring mechanism, and a playback queuing mechanism. The under-run forecasting mechanism determines an estimate of when a supply of data packets to convert will be exhausted. The statistics monitoring mechanism measures the arrival fluctuations of the supply of data packets. The playback queuing mechanism builds latency in the supply of data packets based on the estimate determined by the under-run forecasting mechanism and the arrival fluctuations measured by the statistics monitoring mechanism. In one embodiment, the supply of data packets relates to audio.

Abstract: Briefly, in accordance with one embodiment of the invention, a method for actively characterizing the latency of an audio channel of a computer, such as a personal computer, is provided. At least two signal streams for a waveform are created in the audio channel. The presence of the first signal sample stream for the waveform and the second signal sample stream for the waveform is detected at a point in the audio channel. The time between the detections of the signal sample streams is measured.

Abstract: The present invention guides a talker into a narrow sensitivity region by providing a light that is only visible when the talker's eyes are just above the sensitivity region of a microphone. When the talker keeps the light within his sight while speaking, there is no wavering problem. If the talker cannot see the light, then he is outside the sensitivity region and is alerted to a potential wavering problem by not seeing the light. In this way, the present invention takes advantage of the fact that the talker's eyes are located in close proximity to his mouth. In addition, high frequencies emanating from the mouth are highly directional and applications with speech input, such as speech recognition, function better when these high frequencies are available for analysis.

Abstract: A method and apparatus to perform speech recognition over a voice channel is described.

Abstract: A half duplex switching device includes an input connection for receiving an input audio signal, and classification module coupled to the input connection. The classification module provides an output which indicates a classification of the input signal based upon a density of the input audio signal, an energy level of the input audio signal, and classification data provided with the input audio signal. A switching device is coupled to the classification module and determines if the received input audio signal contains speech signals based upon the output of the classification module. The communication receiving device can be used in both communication systems which provide continuous speech signals, and communication systems which remove silence and only provide speech signals.

Abstract: A half duplex switching device includes an input connection for receiving an input audio signal, and classification module coupled to the input connection. The classification module provides an output which indicates a classification of the input signal based upon a density of the input audio signal, an energy level of the input audio signal, and classification data provided with the input audio signal. A switching device is coupled to the classification module and determines if the received input audio signal contains speech signals based upon the output of the classification module. The communication receiving device can be used in both communication systems which provide continuous speech signals, and communication systems which remove silence and only provide speech signals.