Abstract: Aspects of a method and system for multi-band amplitude estimation and gain control in an audio CODEC are provided. In this regard, an audio signal may be filtered and delayed to generate one or more sub-band signals, a gain may be applied to each sub-band signal to generate one or more level adjusted sub-band signals, and the one or more level adjusted signals may be added to a delayed version of the audio signal. The gain applied to a particular one of the one or more sub-band signals may be controlled based on a detected amplitude of a summed signal derived by summing the particular one of the one or more sub-band signals and a corresponding one of the one or more level-adjusted sub-band signals.

Abstract: 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.

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: A central audio hub, comprising an audio switch, a bus matrix, and an audio buffer, is triggered to read audio samples of an audio stream from the audio buffer. The central audio hub routes the audio samples via the bus matrix to one or more surrounding audio modules such as an audio codec and an audio interface communicatively coupled to the central audio hub. The audio stream may be directly from an external application processor or from an external DDR. With the audio stream from the DDR, a DMA controller may fetch the audio samples from the DDR in response to a request received from the audio buffer, and store the fetched audio samples into the audio buffer for routing. The audio switch may be triggered at a determined sampling rate to read the audio samples from the audio buffer utilizing a determined sample format, and to route the audio samples to the surrounding audio modules.

Abstract: Certain aspects of a method and system for a delay locked loop for a rake receiver are disclosed. Aspects of one method may include normalizing a signal power of a first control channel based on a threshold value. A sampling time associated with at least one or more of the following: the first control channel, a second control channel, an on-time control channel, and a data channel, may be adjusted based on a comparison between the normalized signal power of the first control channel and a signal power of the second control channel. The second control channel may be delayed with respect to the first control channel by a particular time period. The first and second control channels may be common pilot control channels (CPICHs). The combined signal power of the first control channel may be normalized based on said threshold value.

Abstract: In a multiple access environment, to improve decoding efficiency for small code blocks by a constituent decoder, multiple code blocks may be concatenated to form a single larger code block. Example embodiments of the present invention concatenate code blocks for turbo codes I and II. Example embodiments of the present invention may also concatenate code blocks for convolutional code. Example embodiments of the present invention also disclose a system to concatenate the code blocks.

Abstract: 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.

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: A method for detecting transmission power reduction in a physical channel including at least a non-enhanced data channel portion, a non-enhanced control channel portion, an enhanced control channel portion and an enhanced data channel portion is provided. The method includes estimating a ratio of a transmission power associated with the enhanced data channel to transmission power associated with the non-enhanced control channel portion, comparing the estimated ratio with a threshold, the threshold being determined based on a format indicator associated with the enhanced data channel portion, and detecting a transmission power reduction in the enhanced data channel portion based on the comparing step.

Abstract: One embodiment includes determining a channel quality prediction error indicative a channel quality for a first time interval. The first time interval includes of a plurality of subframes, and the channel quality prediction error is calculated based on a first channel quality indicator associated with a first sub-frame and a second channel quality indicator associated with a second sub-frame. The first subframe and the second sub-frame are temporally spaced from one another. For example, the first subframe and the second subframe are temporally spaced apart by at least the length of the first time interval. More specifically, the second subframe may be received the first time interval after the first subframe.

Abstract: A system for determining in-phase and quadrature-phase mismatch in a multiple-input, multiple-output (MIMO) communication architecture includes at least one transmitter coupled to at least one receiver and an in-phase (I) signal, quadrature-phase (Q) signal mismatch element configured to receive and Q signal components over at least one communication channel, the I/Q signal mismatch element also configured to provide a signal representing gain imbalance, a signal representing quadrature error and a signal representing I/Q offset.

Abstract: Whether to process a control channel corresponding to a data channel carrying transmitted data based on a re-transmission indicator and a threshold value is determined at the receiver. The re-transmission indicator indicates a number of times the transmitted data has been transmitted. Control information received on the control channel is then selectively processed based on the determining step.

Abstract: Methods and systems for detecting, and controlling power for, an auxiliary microphone are disclosed. Aspects of one method may include a detection block intermittently enabling a bias circuit block to provide a bias signal to determine if an auxiliary microphone may be communicatively coupled to a mobile device. The detection block may process 1-bit digital samples received from the bias circuit bock to determine whether the auxiliary microphone may be communicatively coupled. The detection block may also process the 1-bit digital samples to determine if a button associated with the auxiliary microphone may have been pushed or activated.

Abstract: In a method of detecting a signal, a control channel associated with a physical channel may be decoded to produce at least one decoding metric. A control channel signal on the control channel may then be detected based on the decoding metric.

Abstract: Apparatus and method to process a pilot channel and a synchronization channel to obtain a combined noise estimate of the pilot channel and the synchronization channel that is synchronized to the pilot channel. The combined noise estimation is processed to determine a first noise component for a period when both the pilot channel and the synchronization channel are present. Then, the combined noise estimate is processed to determine a second noise component for a period when only the pilot channel is present. Next, the second noise component is subtracted from the first noise component to derive a noise estimation for the synchronization channel. The technique may be applied to the Common Pilot Channel (CPICH), and the Synchronization Channel (SCH) defined in a 3rd Generation Partnership Project standard specification.

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.