Abstract: A system level automatic gain control (“System AGC”) automatically initializes and controls analog microphone gain in an environment where multiple independent applications simultaneously receive an input from a single analog microphone or microphone array. In one embodiment, the System AGC also prevents those applications from acting to separately control the gain by intercepting external gain control commands and responding to the corresponding application with a corresponding digital gain applied to the input signal from the microphone. Consequently, the System AGC avoids problems relating to oscillations and instability in the microphone gain resulting from multiple applications trying to simultaneously control the gain while preventing each application from adversely affecting the quality of another application's audio capture signal.

Abstract: An acoustic echo cancellation technique. The present adaptive acoustic echo cancellation technique employs a plurality of acoustic echo cancellation filters which use different adaptation techniques which may employ different parameters such as step size, to improve both the adaptation algorithm convergence time and misadjustment over previously known acoustic echo cancellation techniques.

Abstract: Harmonic distortion residual echo suppression (HDRES) technique embodiments are presented which act to suppress the residual echo remaining after a near-end microphone signal has undergone AEC, including harmonic distortion in the signal that was caused by the speaker audio signal playback. In general, an AEC module is employed which suppresses some parts of the speaker audio signal found in a near-end microphone signal and generates an AEC output signal. A HDRES module then inputs the AEC output signal and the speaker audio signal, and suppresses at least a portion of a residual part of the speaker audio signal that was left unsuppressed by the AEC module. This includes at least a portion of the harmonic distortion exhibited in the AEC output signal.

Abstract: Signal detectors are described herein. By way of example, a system for detecting signals can include a microphone signal detector, a loudspeaker signal detector, a signal discriminator and a decision component. When the microphone signal detector detects the presence of a microphone signal, the loudspeaker signal detector detects the presence of a loudspeaker signal and the signal discriminator determines that near-end speech dominates loudspeaker echo, the decision component can confirm the presence of doubletalk. When the microphone signal detector detects the presence of a microphone signal and the signal discriminator determines that near-end speech dominates loudspeaker echo, the decision component confirms the presence of near-end signal.

Abstract: Hybrid echo canceller controllers are described herein. By way of example, a system for controlling an echo canceller can include a signal indicator and an echo canceller controller. The signal indicator can be configured to indicate periods of near-end signal and to indicate periods of echo only with echo-path change in the corrupted signal based at least in part on cross-correlation between two signals associated with the echo canceller. The echo canceller controller can be configured to control the echo canceller according to indications from the signal indicator.

Abstract: Hybrid echo canceller controllers are described herein. By way of example, a system for controlling an echo canceller can include a cross-correlator, a discriminator and an echo canceller controller. The cross-correlator can be configured to produce a cross-correlation based output that facilitates controlling the echo canceller by cross-correlating two signals associated with the echo canceller. The discriminator can be configured to produce a discriminator output that discriminates between near-end signal and echo in a corrupted signal. The echo canceller controller can be configured to control the echo canceller according to the cross-correlation based output and the discriminator output.

Abstract: Cross-correlation based echo canceller controllers are described herein. By way of example, a system for controlling an echo canceller having one or more adaptive filters can include one or more adaptive filter controllers each corresponding to one of the one or more adaptive filters and each configured to halt adaptation of its corresponding adaptive filter according to the cross-correlation of its corresponding corrupted signal and its corresponding error signal of its corresponding adaptive filter.

Abstract: An echo cancellation technique that can process multi-input microphone signals with only a small increase in the overall CPU consumption compared to implementing the algorithm for a single channel microphone signal. Furthermore, the invention provides an architecture that provides for echo cancellation for multiple applications in parallel with only a small increase in CPU consumption compared to a single instance of echo cancellation with a single microphone input and multi-output channel playback.

Abstract: A regression-based residual echo suppression (RES) system and process for suppressing the portion of the microphone signal corresponding to a playback of a speaker audio signal that was not suppressed by an acoustic echo canceller (AEC). In general, a prescribed regression technique is used between a prescribed spectral attribute of multiple past and present, fixed-length, periods (e.g., frames) of the speaker signal and the same spectral attribute of a current period (e.g., frame) of the echo residual in the output of the AEC. This automatically takes into consideration the correlation between the time periods of the speaker signal. The parameters of the regression can be easily tracked using adaptive methods. Multiple applications of RES can be used to produce better results and this system and process can be applied to stereo-RES as well.

Abstract: Harmonic distortion residual echo suppression (HDRES) technique embodiments are presented which act to suppress the residual echo remaining after a near-end microphone signal has undergone AEC, including harmonic distortion in the signal that was caused by the speaker audio signal playback. In general, an AEC module is employed which suppresses some parts of the speaker audio signal found in a near-end microphone signal and generates an AEC output signal. A HDRES module then inputs the AEC output signal and the speaker audio signal, and suppresses at least a portion of a residual part of the speaker audio signal that was left unsuppressed by the AEC module. This includes at least a portion of the harmonic distortion exhibited in the AEC output signal.

Abstract: An acoustic echo cancellation technique. The present adaptive acoustic echo cancellation technique employs a plurality of acoustic echo cancellation filters which use different adaptation techniques which may employ different parameters such as step size, to improve both the adaptation algorithm convergence time and misadjustment over previously known acoustic echo cancellation techniques.

Abstract: The present invention relates to systems and methods that remove echo from a signal via a novel echo cancellation technique that supports arbitrary playback sampling rates. The novel echo cancellation technique transforms a playback signal to a frequency domain representation and converts its sampling rate to a sampling rate of a frequency domain transformed received signal for the appropriate number of frequency bins. This conversion is achieved via an exact or interpolated approached. The re-sampled playback signal transform is then utilized in connection with the received signal transform to remove echo associated with the playback signal from the received signal.

Abstract: An audio processing system is used to process digital audio signals that represent sound emanating from a source that is moving through three dimensional space. The audio processing system has a filter unit that employs infinite impulse response (IIR) filters to filter the audio signals. The IIR filters have filter coefficients that change when the sound source is stationary or moves from one location to the next in the 3D space. To minimize the transient response following a coefficient change, the filter unit initializes elements in the tap delay lines of the IIR filters to non-zero values. In one implementation, the tap delay line elements are changed to a set of predetermined non-zero values. In another implementation, the tap delay line elements are initialized to the final values produced by the filter for the previous sound source location.