摘要:
A method to perform adaptive channel filtering on a Radio Frequency (RF) bursts in a cellular wireless communication system. This method first filters an input signal with a first stage filter having a first bandwidth to produce a first stage output signal. Then the first stage output signal is filtered with a second stage filter having a second bandwidth narrower than that of the first stage filter to produce a multi-stage output signal. A comparison between first stage performance measurements and multi-stage performance measurements determine the mode of operation of the adaptive multistage filter. A first mode of operation, selected when the first stage performance measurement compares favorably with the second stage performance measurement, selects the output of the first stage filter as the output of the multi-stage filter. Otherwise, a second mode of operation selects the output of the second stage filter as the output of the multi-stage filter.
摘要:
A technique for time tracking helps a mobile communication device with multiple SIMs to more accurately maintain synchronization with a base station. By utilizing synchronization information from both SIMs, the technique is able to more frequently and more accurately adjust timing information for each SIM. As a result, the mobile communication device exhibits an increased ability to accurately synchronize without the need for a higher precision reference or increased power consumption.
摘要:
Disclosed are various embodiments providing adaptive path selection for interference cancellation for wireless communication devices. Signal strength metrics are obtained for each of multiple signal paths. One or more of the signal paths are selected as cancellation candidates in response to determining that the signal paths are associated with a strong interfering path based at least in part on the signal strength metrics for the signal paths and threshold criteria. Cancellation is enabled for an estimated signal generated using the signal paths in response to the signal paths being selected as cancellation candidates.
摘要:
Aspects of a method and system for detecting and identifying electronic accessories or peripherals utilizing a hardware audio CODEC are provided. In this regard, a hardware audio CODEC may be operable to compare one or more voltages on one or more biased pins of an accessory or peripheral port to one or more reference voltages and generate one or more digital representations of the one or more voltages on the biased one or more pins. An accessory or peripheral attached to the accessory or peripheral port may be identified based on the comparison and/or the generated one or more digital representations. The one or more bias voltages may be controlled based on a result of the comparison and/or the generated digital representations. The one or more bias voltages may be reduced after an attached accessory or peripheral has been identified.
摘要:
Aspects of a method and system for decoding single antenna interference cancellation (SAIC) and redundancy processing adaptation using burst process are provided. A wireless receiver may decode bit sequences based on a first decoding algorithm that may utilize redundancy in the data and that may impose physical constraints. The receiver may also decode a received bit sequence based on a second decoding algorithm that utilizes SAIC. Received data may be processed in a burst process portion in either decoding algorithm. Burst processed data from one of the decoding algorithms may be selected based on signal-to-noise ratio and/or received signal level measurements. The selected burst processed data may be communicated to a frame processing portion of the corresponding decoding algorithm.
摘要:
Aspects of a method and system for interference suppression in WCDMA systems may include one or more circuits that are operable to receive a plurality of multipath signals via one or more receiving antennas. A plurality of weighting factor values may be computed based on the received multipath signals. Estimated signals may be based on the weighting factor values. Residual signals may be generated based on received signals and the estimated signals. Addback signals may be generated based on the estimated signals and the residual signals. Updated estimated signals may be generated based on the addback signals and the weighting factor values. Incremental signals may be generated based on the updated estimated signals and addback signals. Updated residual signals may be generated based on the incremental signals and previous residual signals. The interference suppressed signals may be generated based on the updated residual signals and updated estimated signals.
摘要:
In a method and system for audio level detection and control, an amplitude of an audio signal may be compared to a threshold and an attenuation applied to the audio signal may be adjusted based on the comparison. In instances that the amplitude of the audio signal is greater than or equal to the threshold the adjustment may comprise increasing a first attenuation factor until the amplitude of the audio signal is less than the threshold. The first attenuation factor may be subsequently decreased until the amplitude of the audio signal is greater than or equal to the threshold or until the first attenuation factor is equal to zero. The attenuation of the audio signal may be controlled via a digital gain circuit within the hardware audio CODEC, wherein an overall attenuation factor of the digital gain circuit is a sum of the first attenuation factor and a second attenuation factor.
摘要:
Multi-channel noise suppression systems and methods are described that omit the traditional delay-and-sum fixed beamformer in devices that include a primary speech microphone and at least one noise reference microphone with the desired speech being in the near-field of the device. The multi-channel noise suppression systems and methods use a blocking matrix (BM) to remove desired speech in the input speech signal received by the noise reference microphone to get a “cleaner” background noise component. Then, an adaptive noise canceler (ANC) is used to remove the background noise in the input speech signal received by the primary speech microphone based on the “cleaner” background noise component to achieve noise suppression. The filters implemented by the BM and ANC are derived using closed-form solutions that require calculation of time-varying statistics of complex frequency domain signals in the noise suppression system.
摘要:
A method and system for decoding control data in GSM-based systems using inherent redundancy and physical constraints are presented. At least one estimated GSM-based bit sequence may be selected by performing searches that start from trellis junctions determined by the decoding algorithm. The estimated bit sequences may be selected based on corresponding redundancy verification parameters. At least one physical constraint test may be performed on the selected estimated GSM-based bit sequences to select a decoded output GSM-based bit sequence. A multilayer decoding process may comprise a burst process and a frame process. Results from a first burst process may be utilized to generate a decoded GSM bit sequence in the frame process. The frame process may utilize redundancy information and physical constraints to improve the performance of a decoding algorithm.
摘要:
An audio codec in a baseband processor may be utilized for mixing audio signals received at a plurality of data sampling rates. The mixed audio signals may be up sampled to a very large sampling rate, and then down sampled to a specified sampling rate that is compatible with a Bluetooth-enabled device by utilizing an interpolator in the audio codec. The down-sampled signals may be communicated to Bluetooth-enabled devices, such as Bluetooth headsets, or Bluetooth-enabled devices with a USB interface. The interpolator may be a linear interpolator for which the audio codec may enable generation of triggering and/or coefficient signals based on the specified output sampling rate. An interpolation coefficient may be generated based on a base value associated with the specified output sampling rate. The audio codec may enable selecting the specified output sampling rate from a plurality of rates.