Abstract: An apparatus and method for communications is disclosed. The apparatus includes a receiver configured to scale an audio signal, and a transducer circuit configured to provide an ambient signal in response to an ambient condition, wherein the receiver is further configured to scale the ambient signal from the transducer circuit and combine the scaled ambient signal with the scaled audio signal, the receiver being further configured to adjust the scaling applied to at least one of the ambient and audio signals.

Abstract: Certain aspects of the present disclosure relate to a method for quantizing signals and reconstructing signals, and/or encoding or decoding data for storage or transmission. Points of a signal may be determined as local extrema or points where an absolute rise of the signal is greater than a threshold. The tread and value of the points may be quantized, and certain of the quantizations may be discarded before the quantizations are transmitted. After being received, the signal may be reconstructed from the quantizations using an iterative process.

Abstract: Methods and apparatus are described for improving the transmission of multimedia data over wireless communication channels. These techniques include determining a physical layer packet size of the wireless communication system and determining a maximum size of a compressed header. Then, partitioning an information unit, wherein the size of the partitions are selected such that after a partition is encoded the aggregate size of the encoded partition and the compressed header are the size of the physical layer packet, or less. The techniques can be used for various types of information units, such as multimedia data, variable bit rate data streams, video streams, video teleconference stream, or voice over IP.

Abstract: A technique of spectral noise shaping in an audio coding system is disclosed. Frequency decomposition of an input audio signal is performed to obtain multiple frequency sub-bands that closely follow critical bands of human auditory system decomposition. The tonality of each sub-band is determined. If a sub-band is tonal, time domain linear prediction (TDLP) processing is applied to the sub-band, yielding a residual signal and linear predictive coding (LPC) coefficients of an all-pole model representing the sub-band signal. The residual signal is further processed using a frequency domain linear prediction (FDLP) method. The FDLP parameters and LPC coefficients are transferred to a decoder. At the decoder, an inverse-FDLP process is applied to the encoded residual signal followed by an inverse TDLP process, which shapes the quantization noise according to the power spectral density of the original sub-band signal. Non-tonal sub-band signals bypass the TDLP process.

Abstract: A system, method, apparatus, signal-bearing medium, and means for transmitting speech activity in a distributed voice recognition (VR) system. The distributed voice recognition system includes a local VR engine in a subscriber unit (102) and a server VR engine on a server (160). The local VR engine comprises a voice activity detection (VAD) module (106) that detects voice activity within a speech signal, and comprises an advanced feature extraction (AFE) module (104) that extracts features from a speech signal. The detected voice activity information is transmitted over a first wireless communication channel to the server (160). The feature extraction information is transmitted over a second wireless communication channel, separate from the first wireless communication channel, to the server (160). The server (160) processes the received information to determine a linguistic estimate of the electrical speech signal, and transmits the linguistic estimate to the subscriber unit (102).

Abstract: In an apparatus and method, time-varying signals are processed and encoded via a frequency domain linear prediction (FDLP) scheme to arrive at an all-pole model. Residual signals resulted from the scheme are estimated. Quantized values of the all-pole model and the residual signals are packetized as encoded signals suitable for transmission or storage. To reconstruct the time-varying signals, the encoded signals are decoded. The decoding process is basically the reverse of the encoding process.

Abstract: An apparatus and method for communications are disclosed. The apparatus may include an a quantizer having three levels, and a switching power amplifier configured to drive a load having first and second terminals, wherein the switching power amplifier is further configured to switch the first and second terminals between first and second power rails only if the output from the quantizer is at one of the three levels.

Abstract: An apparatus and method for communications are disclosed. The apparatus may include an a quantizer having three levels, and a switching power amplifier configured to drive a load having first and second terminals, wherein the switching power amplifier is further configured to switch the first and second terminals between first and second power rails only if the output from the quantizer is at one of the three levels.

Abstract: A system and method for extracting acoustic features and speech activity on a device and transmitting them in a distributed voice recognition system. The distributed voice recognition system includes a local VR engine in a subscriber unit and a server VR engine on a server. The local VR engine comprises a feature extraction (FE) module that extracts features from a speech signal, and a voice activity detection module (VAD) that detects voice activity within a speech signal. The system includes filters, framing and windowing modules, power spectrum analyzers, a neural network, a nonlinear element, and other components to selectively provide an advanced front end vector including predetermined portions of the voice activity detection indication and extracted features from the subscriber unit to the server. The system also includes a module to generate additional feature vectors on the server from the received features using a feed-forward multilayer perceptron (MLP) and providing the same to the speech server.

Abstract: Methods and apparatus for audio and speech processing including generating a plurality of frames, each of the frames comprising a plurality of transform coefficients, and allocating bits to the transform coefficients in each of the frames such that at least two of the transform coefficients in the same frame have different bit allocations and the total number of the bits allocated to the transform coefficients in at least two of the frames is equal.

Abstract: Certain aspects of the present disclosure relate to a method for compressed sensing (CS). The CS is a signal processing concept wherein significantly fewer sensor measurements than that suggested by Shannon/Nyquist sampling theorem can be used to recover signals with arbitrarily fine resolution. In this disclosure, the CS framework is applied for sensor signal processing in order to support low power robust sensors and reliable communication in Body Area Networks (BANs) for healthcare and fitness applications.

Abstract: Certain aspects of the present disclosure relate to a method for compressed sensing (CS). The CS is a signal processing concept wherein significantly fewer sensor measurements than that suggested by Shannon/Nyquist sampling theorem can be used to recover signals with arbitrarily fine resolution. In this disclosure, the CS framework is applied for sensor signal processing in order to support low power robust sensors and reliable communication in Body Area Networks (BANs) for healthcare and fitness applications.

Abstract: A system and method for transmitting speech activity in a distributed voice recognition system. The distributed voice recognition system includes a local VR engine in a subscriber unit and a server VR engine on a server. The local VR engine comprises a feature extraction (FE) module that extracts features from a speech signal, and a voice activity detection module (VAD) that detects voice activity within a speech signal. Indications of voice activity are transmitted ahead of features from the subscriber unit to the server.

Abstract: Certain aspects of the present disclosure relate to a technique for mitigating artifacts of biophysical signals in a body area network. Information from multiple sensors (including motion information of the body) can be employed in mitigating the artifacts. The biophysical signals in the body area network can be compressively sensed.

Abstract: Certain aspects of the present disclosure relate to a method for compressed sensing (CS). The CS is a signal processing concept wherein significantly fewer sensor measurements than that suggested by Shannon/Nyquist sampling theorem can be used to recover signals with arbitrarily fine resolution. In this disclosure, the CS framework is applied for sensor signal processing in order to support low power robust sensors and reliable communication in Body Area Networks (BANs) for healthcare and fitness applications.

Abstract: A method and apparatus for audio signal processing by applying log companding on spectral domain or time domain representations of the audio signals to provide an encoded audio signal, which is decoded upon receipt. A frequency domain representation or time domain representation of the audio signal is computed by separating the audio signal into specific frequency bands, each having a coefficient. Log companding with different compression ratios is performed on each coefficient to provide an encoded signal. Upon receipt of the encoded signal, inverse log companding and time frequency or time scale reconstruction are performed to provide the audio signal.

Abstract: A system and method for media access control are disclosed. The method comprises providing concurrent orthogonal channels to access media using pulse division multiple access to define pulse positions, wherein the pulse division multiple access includes a time hopping sequence and an offset to distinguish the concurrent orthogonal channels. In addition, the method comprises processing signals associated with at least one of the orthogonal channels.

Abstract: Certain aspects of the present disclosure relate to a method for estimating a blood pressure using both a pulse arrival time (PAT) and an instantaneous heart rate (HR). The PAT can be measured as the delay between QRS peaks in an electrocardiogram (ECG) signal and corresponding points in a photoplethysmogram (PPG) waveform. Parameters of the estimation model can be determined through an initial training. Then, the model parameters can be recalibrated in constant intervals using the recursive least square (RLS) approach combined with a smooth bias fixing. The proposed estimation algorithm is applied on a multi-parameter intelligent monitoring for intensive care (MIMIC) database, and the results are compared with estimation methods that use PAT only or HR only. The proposed estimation algorithm meets, on average, the Association for the Advancement of Medical Instrumentation (AAMI) requirements and outperforms other methods from the prior art.

Abstract: An apparatus and method for processing data are disclosed. The apparatus may include a receiver clock, and a processing system configured to use the receiver clock to receive data from a transmitter, the data being generated with a transmitter clock in the transmitter, wherein the processing system is further configured to estimate a mismatch between the transmitter and receiver clocks, and to determine whether to modify the data based on the estimated mismatch.

Abstract: Certain aspects of the present disclosure relate to a method for compressed sensing (CS). The CS is a signal processing concept wherein significantly fewer sensor measurements than that suggested by Shannon/Nyquist sampling theorem can be used to recover signals with arbitrarily fine resolution. In this disclosure, the CS framework is applied for sensor signal processing in order to support low power robust sensors and reliable communication in Body Area Networks (BANs) for healthcare and fitness applications.