| Patent application number | Description | Published |
|---|
| 20080267288 | METHOD, APPARATUS, AND SYSTEM FOR ENHANCING ROBUSTNESS OF PREDICTIVE VIDEO CODECS USING A SIDE-CHANNEL BASED ON DISTRIBUTED SOURCE CODING TECHNIQUES - A method, apparatus, and system for providing distributed source coding techniques that improve data coding performance, such as video data coding, when channel errors or losses occur. Errors in the reconstruction of the data is eliminated or reduced by sending extra information. Correlation between a predicted sequence and an original sequence can be used to design codebooks and find the cosets required to represent the original image. This information may be sent over another channel, or a secondary channel. | 10-30-2008 |
| 20090021408 | ADAPTIVE DYNAMIC RANGE CONTROL - Apparatus and method for processing signals. A sigma-delta modulator is used. An adaptive dynamic range controller is configured to adaptively adjust the dynamic range of a signal output from the sigma-delta modulator. | 01-22-2009 |
| 20090021572 | CONTENT- AND LINK-DEPENDENT CODING ADAPTATION FOR MULTIMEDIA TELEPHONY - This disclosure describes techniques that can facilitate multimedia telephony. In one example, a method for communication of multimedia data comprises determining a first level of throughput associated with multimedia data communication from a first access terminal to a network, determining a second level of throughput associated with multimedia data communication from the network to a second access terminal based on feedback from the second access terminal to the first access terminal via the network, determining a budget associated with communication of a video unit of the multimedia data, and coding the video unit of the multimedia data based on the budget and the first and second levels of throughput. | 01-22-2009 |
| 20090198500 | TEMPORAL MASKING IN AUDIO CODING BASED ON SPECTRAL DYNAMICS IN FREQUENCY SUB-BANDS - An audio coding technique based on modeling spectral dynamics 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. Each sub-band is then frequency transformed and linear prediction is applied. This results in a Hilbert envelope and a Hilbert Carrier for each of the sub-bands. Because of application of linear prediction to frequency components, the technique is called Frequency Domain Linear Prediction (FDLP). The Hilbert envelope and the Hilbert Carrier are analogous to spectral envelope and excitation signals in the Time Domain Linear Prediction (TDLP) techniques. Temporal masking is applied to the FDLP sub-bands to improve the compression efficiency. Specifically, forward masking of the sub-band FDLP carrier signal can be employed to improve compression efficiency of an encoded signal. | 08-06-2009 |
| 20090259671 | SYNCHRONIZING TIMING MISMATCH BY DATA INSERTION - The rate at which data is provided by one device and the rate at which that data is processed by another device may differ. For example, a transmitting device may transmit data according to a transmit clock while a receiving device that receives the transmitted data may process the data according to a receive clock. If there is a timing mismatch between the transmit and receive clocks, the receiving device may receive data faster or slower than it processes the data. In such a case, there may be errors relating to the processing of the received data. To address timing mismatches such as this, the receiving device may delete data from or insert data into the received data. In conjunction with these operations, the receiving device may modify the received data at or near the insertion point or the deletion point in a manner that mitigates any adverse effect the insertion or deletion may have on a resulting output signal. | 10-15-2009 |
| 20090259672 | SYNCHRONIZING TIMING MISMATCH BY DATA DELETION - The rate at which data is provided by one device and the rate at which that data is processed by another device may differ. For example, a transmitting device may transmit data according to a transmit clock while a receiving device that receives the transmitted data may process the data according to a receive clock. If there is a timing mismatch between the transmit and receive clocks, the receiving device may receive data faster or slower than it processes the data. In such a case, there may be errors relating to the processing of the received data. To address timing mismatches such as this, the receiving device may delete data from or insert data into the received data. In conjunction with these operations, the receiving device may modify the received data at or near the insertion point or the deletion point in a manner that mitigates any adverse effect the insertion or deletion may have on a resulting output signal. | 10-15-2009 |
| 20090259906 | DATA SUBSTITUTION SCHEME FOR OVERSAMPLED DATA - Low latency and computationally efficient techniques may be employed to account for errors in data such as low bit-width, oversampled data. In some aspects these techniques may be employed to mitigate audio artifacts associated with sigma-delta modulated audio data. In some aspects an error may be detected in a set of encoded data based on an outcome of a channel decoding process. Upon determining that a set of data may contain at least one error, the set of data may be replaced with another set of data that is based on one or more neighboring data sets. For example, in some aspects a set of data including at least one bit in error may be replaced with data that is generated by applying a cross-fading operation to neighboring data sets. In some aspects a given data bit may be flipped as a result of a linear prediction operation that is applied to PCM equivalent data that is associated with the given data bit and its neighboring data bits. In some aspects a set of data including at least one bit in error may be replaced with data that is generated by performing linear interpolation operations on PCM equivalent data that is associated with neighboring data sets. | 10-15-2009 |
| 20090259922 | CHANNEL DECODING-BASED ERROR DETECTION - Low latency and computationally efficient techniques may be employed to account for errors in data such as low bit-width, oversampled data. In some aspects these techniques may be employed to mitigate audio artifacts associated with sigma-delta modulated audio data. In some aspects an error may be detected in a set of encoded data based on an outcome of a channel decoding process. Upon determining that a set of data may contain at least one error, the set of data may be replaced with another set of data that is based on one or more neighboring data sets. For example, in some aspects a set of data including at least one bit in error may be replaced with data that is generated by applying a cross-fading operation to neighboring data sets. In some aspects a given data bit may be flipped as a result of a linear prediction operation that is applied to PCM equivalent data that is associated with the given data bit and its neighboring data bits. In some aspects a set of data including at least one bit in error may be replaced with data that is generated by performing linear interpolation operations on PCM equivalent data that is associated with neighboring data sets. | 10-15-2009 |
| 20090323985 | SYSTEM AND METHOD OF CONTROLLING POWER CONSUMPTION IN RESPONSE TO VOLUME CONTROL - An apparatus for audio processing including a first device (e.g., a multiplier, digital signal gain module, etc.) adapted to apply a gain to a first digital audio signal to generate a second digital audio signal; a second device (e.g., a digital-to-analog converter (DAC), etc.) adapted to generate an analog audio signal from the second digital audio signal; a third device (e.g., a detector, sensor, user interface, etc.) adapted to generate an audio characteristic signal related to a characteristic of the first or second digital audio signal, or the analog audio signal; and a fourth device (e.g., a controller, control module, etc.) adapted to control the gain of the first device based on a first function of the audio characteristic signal, and control a power supplied to the second device based on a second function of the audio characteristic signal. | 12-31-2009 |
| 20100019845 | SWITCHING POWER AMPLIFIER FOR QUANTIZED SIGNALS - 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. | 01-28-2010 |
| 20100020978 | METHOD AND APPARATUS FOR RENDERING AMBIENT SIGNALS - 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. | 01-28-2010 |
| 20100020985 | METHOD AND APPARATUS FOR REDUCING AUDIO ARTIFACTS - An apparatus and method for processing signals are disclosed. The apparatus may include a receiver configured to receive an audio signal having a plurality of audio artifacts, and an audio circuit configured to reduce the audio artifacts during at least a portion of a time period as a function of an energy level of the audio signal during that time period. | 01-28-2010 |
| 20100023142 | METHOD AND APPARATUS FOR TRANSMIT AND RECEIVE CLOCK MISMATCH COMPENSATION - An apparatus and method for processing signals are disclosed. The apparatus may include an oversampling circuit configured to receive a plurality of audio signal samples, the oversampling circuit being further configured to replicate each of the audio signal samples n times, wherein n is variable. | 01-28-2010 |
| 20100080331 | METHOD AND APPARATUS FOR INTEGRATED CLOCK MISMATCH COMPENSATION AND PACKET LOSS CONCEALMENT - 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. | 04-01-2010 |
| 20100081946 | METHOD AND APPARATUS FOR NON-INVASIVE CUFF-LESS BLOOD PRESSURE ESTIMATION USING PULSE ARRIVAL TIME AND HEART RATE WITH ADAPTIVE CALIBRATION - 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. It is also shown in the present disclosure that the proposed estimation algorithm is robust to unknown skew between the ECG and PPG signals. | 04-01-2010 |
| 20100082302 | METHOD AND APPARATUS FOR UNDER-SAMPLED ACQUISITION AND TRANSMISSION OF PHOTOPLETHYSMOGRAPH (PPG) DATA AND RECONSTRUCTION OF FULL BAND PPG DATA AT THE RECEIVER - 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. | 04-01-2010 |
| 20100086073 | SYSTEM AND METHOD TO IMPLEMENT CONCURRENT ORTHOGONAL CHANNELS IN AN ULTRA-WIDE BAND WIRELESS COMMUNICATIONS NETWORK - 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. | 04-08-2010 |
| 20100106269 | METHOD AND APPARATUS FOR SIGNAL PROCESSING USING TRANSFORM-DOMAIN LOG-COMPANDING - 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. | 04-29-2010 |
| 20100246651 | PACKET LOSS MITIGATION IN TRANSMISSION OF BIOMEDICAL SIGNALS FOR HEALTHCARE AND FITNESS APPLICATIONS - 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. | 09-30-2010 |
| 20110066381 | METHOD AND APPARATUS FOR ARTIFACTS MITIGATION WITH MULTIPLE WIRELESS SENSORS - 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. | 03-17-2011 |
| 20110134906 | METHOD AND APPARATUS FOR DISTRIBUTED PROCESSING FOR WIRELESS SENSORS - 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. | 06-09-2011 |
| 20110136536 | METHOD AND APPARATUS FOR DISTRIBUTED PROCESSING FOR WIRELESS SENSORS - 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. | 06-09-2011 |
| 20110153315 | AUDIO AND SPEECH PROCESSING WITH OPTIMAL BIT-ALLOCATION FOR CONSTANT BIT RATE APPLICATIONS - 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. | 06-23-2011 |
| 20110153326 | SYSTEM AND METHOD FOR COMPUTING AND TRANSMITTING PARAMETERS IN A DISTRIBUTED VOICE RECOGNITION SYSTEM - 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. | 06-23-2011 |
