Methods and apparatus for a communication system having microphones and loudspeakers to determine a noise and speech level estimate for a transformed signal, determine a SNR from the noise and speech level estimates, and determine a gain for the transformed signal to achieve a selected SNR range at a given position. In one embodiment, the gain is determined by adapting an actual gain to follow a target gain, wherein the target gain is adjusted to achieve the selected SNR range.

An automatic gain control system for a receiver, including: an automatic gain control loop (40) adapted to be coupled to both a first transimpedance amplifier (12) coupled to a first analog-to-digital converter (14) forming a first tributary and a second transimpedance amplifier (12) coupled to a second analog-to-digital converter (14) forming a second tributary; and an offset gain control voltage to gain balance a transimpedance amplifier gain of the first tributary and a transimpedance amplifier gain of the second tributary. The automatic gain control loop can be analog. Also, the automatic gain control loop can be implemented in hardware or firmware.

The invention provides an efficient implementation of cross-product enhanced high-frequency reconstruction (HFR), wherein a new component at frequency QΩ+rΩ.sub.0 is generated on the basis of existing components at Ω and Ω+Ω.sub.0. The invention provides a block-based harmonic transposition, wherein a time block of complex subband samples is processed with a common phase modification. Superposition of several modified samples has the net effect of limiting undesirable intermodulation products, thereby enabling a coarser frequency resolution and/or lower degree of oversampling to be used. In one embodiment, the invention further includes a window function suitable for use with block-based cross-product enhanced HFR. A hardware embodiment of the invention may include an analysis filter bank, a subband processing unit configurable by control data and a synthesis filter bank.

A microphone may comprise a microphone element for detecting sound, and a digital signal processor configured to process a first audio signal that is based on the sound in accordance with a selected one of a plurality of digital signal processing (DSP) modes. Each of the DSP modes may be for processing the first audio signal in a different way. For example, the DSP modes may account for distance of the person speaking (e.g., near versus far) and/or desired tone (e.g., darker, neutral, or bright tone). At least some of the modes may have, for example, an automatic level control setting to provide a more consistent volume as the user changes their distance from the microphone or changes their speaking level, and that may be associated with particular default (and/or adjustable) values of the parameters attack, hold, decay, maximum gain, and/or target gain, each depending upon which DSP is being applied.

A logarithmic amplifier includes programmable gain amplifiers each having a different gain, wherein an input of each of the programmable gain amplifiers is coupled to an input of the logarithmic amplifier; and a summing circuit having inputs coupled to a corresponding output of each of the programmable gain amplifiers and an output coupled to an output of the logarithmic amplifier, wherein the summing circuit generates a logarithmic transfer function having piecewise linear segments.

In one embodiment, an apparatus includes: a low noise amplifier (LNA) to receive and amplify a radio frequency (RF) signal, the LNA having a first controllable gain; a mixer to downconvert the RF signal to a second frequency signal; a programmable gain amplifier (PGA) coupled to the mixer to amplify the second frequency signal, the PGA having a second controllable gain; a digitizer to digitize the second frequency signal to a digitized signal; a demodulator coupled to the digitizer to demodulate the digitized signal; an automatic gain control (AGC) circuit to control one or more of the first controllable gain and the second controllable gain; and an AGC settling circuit to cause the demodulator to begin operation in response to determining that the AGC circuit has settled.

Many portable playback devices cannot decode and playback encoded audio content having wide bandwidth and wide dynamic range with consistent loudness and intelligibility unless the encoded audio content has been prepared specially for these devices. This problem can be overcome by including with the encoded content some metadata that specifies a suitable dynamic range compression profile by either absolute values or differential values relative to another known compression profile. A playback device may also adaptively apply gain and limiting to the playback audio. Implementations in encoders, in transcoders and in decoders are disclosed.

This application relates to Class D amplifier circuits. A modulator controls a Class D output stage based on a modulator input signal (Dm) to generate an output signal (Vout) which is representative of an input signal (Din). An error block, which may comprise an ADC, generates an error signal (ε) from the output signal and the input signal. In various embodiments the extent to which the error signal (ε) contributes to the modulator input signal (Dm) is variable based on an indication of the amplitude of the input signal (Din). The error signal may be received at a first input of a signal selector block. The input signal may be received at a second input of the signal selector block. The signal selector block may be operable in first and second modes of operation, wherein in the first mode the modulator input signal is based at least in part on the error signal; and in the second mode the modulator input signal is based on the digital input signal and is independent of the error signal. The error signal can be used to reduce distortion at high signal levels but is not used at low signal levels and so the noise floor at low signal levels does not depend on the component of the error block.

Apparatus, systems, articles of manufacture, and methods for volume adjustment are disclosed herein. An example method includes collecting data corresponding to a volume of an audio signal as the audio signal is output through a device, when an average volume of the audio signal does not satisfy a volume threshold for a specified timespan, determining a difference between the average volume and a desired volume, and applying a gain to the audio signal to adjust the volume of the audio signal to the desired volume, the gain determined based on the difference between the average volume and the desired volume.

A method for measuring a received signal includes receiving a differential pair of signals by a differential pair of input nodes of a differential circuit. The method includes attempting to match a first current through a first node of the differential circuit corresponding to the differential pair of signals to a second current through a second node of the differential circuit corresponding to a feedback signal. The method includes generating an output measurement signal based on the first current and the second current. The output measurement signal has a level corresponding to an average amplitude of the differential pair of signals.