Abstract: A method of determining handoff of a user equipment (UE) to a transferee serving station from a transferor serving station at the transferee serving station is provided. The transferee serving station (e.g., a Node B) measures an energy level in a first portion of an uplink channel (e.g., an uplink high-speed dedicated physical control channel (HS-DPCCH)), the first portion of the uplink channel associated with the transferee serving station. In an example, the first portion of the uplink channel is where a channel quality indicator (CQI) is expected to be reported from the UE for the transferee serving station. The transferee serving station also measures an energy level in a second portion of the uplink channel, the second portion of the uplink channel associated with the transferor serving station. In an example, the second portion of the uplink channel is where a CQI is expected to be reported from the UE for the transferor serving station.

Abstract: Methods and systems for processing audio signals are disclosed herein. Aspects of the method may comprise controlling gain during multipath, multi-rate audio processing by generating a digital signal that is a product of an input digital signal and a gain coefficient derived from a lookup table, and coarse tuning the gain by bit-shifting the digital signal to generate a digital output signal. The gain may be fine-tuned utilizing a variable step size determined by interpolation. The gain coefficient may be partitioned into gain blocks, which may be twice a corresponding value in preceding gain blocks. The gain blocks may be partitioned into steps that represent particular gain values within a range associated with the gain block. The digital output signal may be ramped utilizing a linear interpolation of the gain coefficients over a number of samples of the input signal, where the number of samples is given as a power of two.

Abstract: Methods and systems for audio CODEC voice ADC processing are disclosed. Aspects of one method may include using a decimating filter that may be enabled to generate 13 MHz, 9-level digital output signal from a 26 MHz, 3-level digital input signal. The 13 MHz, 9-level digital output signal may be processed for RF transmission, for audio output to an output device, and/or utilized for testing by a test fixture, for example. The 13 MHz, 9-level digital output signal may be further processed to generate a 6.5 MHz, 33-level digital signal. The 6.5 MHz, 33-level digital signal may be converted to an analog signal, and processed for audio output and/or testing. The 13 MHz, 9-level digital output signal may also be processed to generate a 40 KHz, 17-bit digital signal, which may be communicated to a test equipment or further processed for RF transmission.

Abstract: 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.

Abstract: Methods and systems for processing audio signals are disclosed herein. Aspects of the method may comprise controlling gain during multipath, multi-rate audio processing by generating via a digital gain circuit, a digital signal that is a product of an input digital signal and a gain coefficient derived from a lookup table, and bit-shifted to generate a digital output signal. The gain coefficient may be partitioned into a number of gain blocks. The gain values in each of the gain blocks may be twice a corresponding value in each preceding gain block. The gain blocks may be partitioned into steps that represent particular gain values within a range associated with the gain block. The digital output signal may be ramped utilizing a linear interpolation of the gain coefficients one step apart over a number of samples of the digital input signal, where the number of samples is given as a power of two.

Abstract: A method to adapt channel quality indicator (CQI) reports from user equipment (UE) is provided. This involves first determining the presence of a high-speed shared control channel (HS-SCCH) signal. An estimated signal to noise ratio (SNR) is also determined. Next an SNR correction based on the presence or lack thereof of the HS-SCCH signal and the CRC checks for HS-SCCH and HS-DSCH signals are determined and applied to the estimated SNR. The CQI report is then generated based on the corrected estimated SNR. This CQI report which takes into account a corrected estimated SNR may then be used to adjust the HS-SCCH signal in an HSDPA telephony system.

Abstract: Methods and systems for audio CODEC voice ADC processing are disclosed. Aspects of one method may include using a decimating filter that may be enabled to generate 13 MHz, 9-level digital output signal from a 26 MHz, 3-level digital input signal. The 13 MHz, 9-level digital output signal may be processed for RF transmission, for audio output to an output device, and/or utilized for testing by a test fixture, for example. The 13 MHz, 9-level digital output signal may be further processed to generate a 6.5 MHz, 33-level digital signal. The 6.5 MHz, 33-level digital signal may be converted to an analog signal, and processed for audio output and/or testing. The 13 MHz, 9-level digital output signal may also be processed to generate a 40 KHz, 17-bit digital signal, which may be communicated to a test equipment or further processed for RF transmission.

Abstract: 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.

Abstract: In a method for power reduction at a receiver, a first radio frame is processed to recover transmitted data based on transport format information. The transport format information is determined based on a combination of transport format information associated with a plurality of received radio frames rather than a single frame. In another method for power reduction at a receiver, whether transport format information associated with a first radio frame is different from transport format information associated with a second radio frame is determined. The transport channel data is then processed based on the determination. The first radio frame is received prior to the second radio frame, and the transport format information indicates a transport format for transport channel data received in the first and second radio frames.

Abstract: A method of scaling soft symbols of an uplink E-DCH is provided, where the E-DCH is received from a user in a network for processing in a base station receiver in the network employing a log-MAP turbo decoding algorithm to process the E-DCH. The E-DCH includes an E-DPDCH from which the soft symbols are generated at the base station receiver, and an E-DPCCH used to transmit control information associated with the E-DPDCH, which along with configuration information from a radio network controller (RNC) of the network enables the base station receiver to determine a reference amplitude ratio linked to the actual power offset of the E-DPDCH from the legacy DPCCH. In the method, an estimated E-DPDCH to DPCCH amplitude ratio that represents a scaling factor for the soft symbols is determined, and the soft symbols are scaled by the scaling factor to enable the soft symbols to be processed by the log-MAP turbo decoding algorithm in the base station receiver.

Abstract: In a method of multipath acquisition for a dedicated traffic channel, path positions of the traffic channel that have desired signal energy for processing in a base station receiver may be determined as a function of information contained in a data part and a control part of the traffic channel.

Abstract: Aspects of a method and system for WCDMA/HSDPA timing adjustment may include adjusting an uplink transmission timing of a signal for transmission by a large adjustment step, utilizing a plurality of smaller adjustment steps, to generate an adjusted transmission signal. The transmission signal may be upsampled to a rate corresponding to the smaller adjustment step, which may generate an upsampled transmission signal. An adjusted upsampled transmission signal may be generated by inserting one or more of the smaller adjustment steps via one or more extra samples into the upsampled transmission signal to generate a delay, or removing one or more samples from the upsampled transmission signal to generate an advance. The adjusted upsampled transmission signal may be filtered and downsampled to generate the adjusted transmission signal.

Abstract: 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.

Abstract: Aspects of a method and system for a delay-locked loop for closely spaced multipath may include determining a difference signal computed from one or more early energies and one or more late energies associated with one or more channel taps selected from a plurality of channel taps. A fat finger timing may be adjusted based on the difference signal, the fat finger comprising the plurality of channel taps that are spaced contiguously at chip period intervals. The one or more early energies and the one or more late energies may be determined based on an offset of Tc/2 or 3Tc/8, where Tc denotes the chip period interval. The offset may be measured from an energy peak associated with a multipath component that may be associated with the fat finger.

Abstract: A first transceiver determines whether a second transceiver has properly decoded at least one of a received data packet and control information associated with the corresponding data packet based on an estimated probability. The estimated probability is indicative of whether the second transceiver has properly decoded at least one of the data packet and associated control information.

Abstract: Apparatus and method to provide unified STTD/CLTD dedicated pilot processing in a wireless receiver. The technique allows different set of parameters to be introduced to a same processing module to process STTD and CLTD diversity signals to recover a pilot signal. Introducing another set of parameters to the processing module also allows processing of a non-diversity signal to recover a pilot. The unified processing of STTD/CLTD signals is achieved by converting STTD/CLTD pilot bits as Hadamard-like bits and processing these bits along with orthogonal pilot bits which are encoded as Hadamard encoded bits.

Abstract: Apparatus and method for processing a received pilot to obtain noise estimation of a pilot channel and processing a received first signal channel, separately from a second signal channel that is orthogonal to the first signal channel, to obtain a signal level of the first signal channel. The noise estimation and the signal level are used to calculate a SNR value. Detection on the first channel is done by checking the SNR and the sign of a soft metric.

Abstract: Aspects of a method and system for UMTS HSDPA Shared Control Channel processing may include calculating at a receiver, for each one of a plurality of control channels, a quality metric based at least one Viterbi Decoder state metric. A control channel may be selected on the basis of the quality metrics, where the quality metric is selected that provides maximum confidence. The selected control channel may be chosen if its corresponding 3GPP metric is greater than a specified threshold, where the threshold is a design parameter. A validity of a selected control channel may be determined based on consistency and a CRC, where the CRC may be derived from decoding a sub-frame. The calculating and selecting may be done for a first slot of a sub-frame for High-Speed Shared Control Channels.

Abstract: A method of multi-path acquisition may include estimating a contribution of acquired users of a first type to a received signal, canceling the estimated contribution from the received signal to generate a resultant signal, and performing multi-path acquisition for users of the first type and users of a second type based on the resultant signal.

Abstract: One or more circuits in a mobile phone may be utilized for up sampling two or more audio signals to a same data sampling rate. Each audio signal, such as digital audio, voice, and polyringer, for example, may be received at one of a plurality of data sampling rates and one or more of the following wireless standards: WCDMA, HSDPA, GSM, GPRS, EDGE, and/or Bluetooth. Audio signals may be equalized and/or compensated with an FIR filter before up sampling or with an IIR filter to reduce overall processing latency. Multiple half-band interpolation operations may perform the up sampling. The first half-band filter may be replaced by an IIR filter to reduce overall processing latency. A gain of the up-sampled data may be adjusted to reduce noise effects. The channels of the up-sampled audio signals may be mixed and later further up sampled for subsequent communication to an output device.