Abstract: An audio compressor may regulate the level of an audio input signal depending on whether the level is above or below a threshold value. The audio compressor may control a pumping that is created when regulating a dynamic range of an audio signal with the threshold value. A feedback loop connecting the signal output from the audio compressor with a release filter may be used to modify an effective release time of the signal. A controller may be used that allows a filter coefficient of the release filter to be controlled to adjust the effective release time as a function of the signal output.

Abstract: A system that models a microphone may include capsules that receive individual signals. The signals may be combined and modified based on a weighting factor. Directivity patterns of a converted signal may be modified or controlled based on the weighting of the signals.

Abstract: The system converts an analog input signal into a digital output signal while avoiding interference, such as clipping. The system derives two signal components having differing signal magnitude levels from the analog input signal. The signal components are subjected to independent analog to digital conversion, such as through low resolution analog to digital converters. The system determines weighting factors for the two signal components based on at least one property of at least one of the signal components, such as the level of at least one of the signal components. The signal components are weighted. The weighted signal components are merged.

Abstract: The system converts an analog input signal into a digital output signal while avoiding interference, such as clipping. The system derives two signal components having differing signal magnitude levels from the analog input signal. The signal components are subjected to independent analog to digital conversion, such as through low resolution analog to digital converters. The system determines weighting factors for the two signal components based on at least one property of at least one of the signal components, such as the level of at least one of the signal components. The signal components are weighted. The weighted signal components are merged.

Abstract: An audio compressor may regulate the level of an audio input signal depending on whether the level is above or below a threshold value. The audio compressor may control a pumping that is created when regulating a dynamic range of an audio signal with the threshold value. A feedback loop connecting the signal output from the audio compressor with a release filter may be used to modify an effective release time of the signal. A controller may be used that allows a filter coefficient of the release filter to be controlled to adjust the effective release time as a function of the signal output.

Abstract: A stereo multiplex encoder (MPX) for FM transmission with a digital signal processor utilizes residual sideband modulation in stereo transmission instead of the conventional double sideband modulation. The adjustment of the sampling frequency to the modulation frequency (fmod=fs/2) is carried out, so that the antialiasing filter of the DAC can be used which produces the residual sideband modulated signal.

Abstract: A system simulates a room impression and/or sound impression to a listener. The simulation includes recreating a sound at a particular location. An audio signal is combined with a space impulse response from that location. The space impulse response and audio signal are split into more than one sub-bands and combined to produce an aural output.

Abstract: A system that models a microphone may include capsules that receive individual signals. The signals may be combined and modified based on a weighting factor. Directivity patterns of a converted signal may be modified or controlled based on the weighting of the signals.

Abstract: A stereo multiplex encoder (MPX) for FM transmission with a digital signal processor utilizes residual sideband modulation in stereo transmission instead of the conventional double sideband modulation. The adjustment of the sampling frequency to the modulation frequency (fmod=fs/2) is carried out, so that the antialiasing filter of the DAC can be used which produces the residual sideband modulated signal.