Abstract: Various embodiments of systems and methods for reducing audio noise are disclosed. One or more sound components such as noise and network tone can be detected based on power spectrum obtained from a time-domain signal. Results of such detection can be used to make decisions in determination of an adjustment spectrum that can be applied to the power spectrum. The adjusted spectrum can be transformed back into a time-domain signal that substantially removes undesirable noise(s) and/or accounts for known sound components such as the network tone.

Abstract: A voice enhancement logic improves the perceptual quality of a processed voice. The voice enhancement system includes a noise detector and a noise attenuator. The noise detector detects a wind buffet and a continuous noise by modeling the wind buffet. The noise attenuator dampens the wind buffet to improve the intelligibility of an unvoiced, a fully voiced, or a mixed voice segment.

Abstract: Systems and methods for minimizing startup transients in digital audio controllers that may result in audible artifacts in the output of an audio amplification system. One embodiment comprises a digital PWM amplifier that includes a mechanism for controlling the amount of dead time in the audio output signal. When the amplifier starts up, the PWM signals provided to the output stage are simultaneously deasserted (i.e., there is dead time) for most of each switch period. The amount of dead time is gradually reduced over a series of switch periods until a nominal operating amount of dead time in each switch period is reached. Thus, the PWM signals are slowly ramped up from having a very large percentage of dead time (e.g., nearly 100%) to having a very small percentage of dead time (e.g., 1-2% to prevent shoot-through.

Abstract: Methods, systems, and apparatus, including computer program products, for restoring audio signals. A data sequence of samples representing an audio signal is received. Multiple filter coefficients are defined for a filter, and a current sample in the data sequence is selected to be processed. The filter coefficients are updated based on a previous sample preceding the current sample in the data sequence and a filtered value determined by the filter for the previous sample. A filtered value for the current sample is determined using the filter with the updated filter coefficients. The filtered value of the current sample is used to determine whether the current sample has been corrupted by impulsive noise, for example, a crackle.

Abstract: Systems and methods for providing protection from failure events in a digital audio amplification system. One embodiment of the invention comprises a system having a digital amplifier controller, an amplifier output stage coupled to the controller and configured to receive audio signals from the controller, one or more sensors coupled to the output stage and one or more low-pass filters coupled to receive sensor signals from the one or more sensors. The low-pass filter is configured to filter the sensor signals and to provide the filtered sensor signals to the controller, which provides a programmable response to the filtered sensor signals. The response may range from not taking any action, to limiting the amplification of audio signals, to shutting down the system.

Abstract: Systems and methods for detecting clipping conditions in an audio signal and processing the signal to reduce the clipping conditions. In one embodiment, a system comprises a noise shaper, a modulator, an output stage and other components. A detector detects clipping in the noise shaper and a signal processor processes the audio signal input to the noise shaper based on feedback received from the detector. The signal processor may function to modify the input audio signal in different ways in response to different conditions that are detected by the detector. A filter may be included to filter the output of the detector before being provided to the signal processor. A flag circuit may be coupled between the filter and the signal processor to assert an output signal until the signal processor resets the flag circuit.

Abstract: There is a problem that a delay is caused in signal processing by exercising control so as to cause amplitude peaks in a transmission signal to fall in a prescribed range. A transmission apparatus for conducting wireless communication makes a decision whether a specific pattern which causes a peak exceeding a predetermined amplitude range to be generated in frequency characteristics of a transmission signal is included in a transmission code sequence for forming the transmission signal. And the transmission apparatus selects filter coefficients which prescribe a band limiting factor for the frequency characteristics of the transmission signal, on the basis of a result of the decision, and conducts filtering on the transmission code sequence by using the selected filter coefficients.

Abstract: A method and system for transmitting and receiving a data stream using a phase or frequency modulated audio signal wherein an audio signal is modulated in at least one of phase and frequency according to the data stream and wherein the modulated audio signal is outside the range of human hearing. The modulated audio signal is combined with an audio program into a composite signal that is used to modulate a carrier. In order to receive the data easily, the audio program is filtered to accommodate the modulated audio signal. During reception, the gain of a detector is controlled by the power level of the recovered data-modulated audio signal. Recovery of the data-modulated audio stream is accomplished by first recovering the composite signal from the modulated carrier and attenuating the audio program so as to isolate the data-modulated audio signal. The isolated data-modulated audio signal is then again demodulated to recover the transmitted data.

Abstract: An apparatus is described that includes an audio power amplifier having an input and an output. An alternating-current to direct-current power converter is coupled to the audio power amplifier in a single package to supply power to the audio power amplifier.

Abstract: An audio control apparatus. The audio control apparatus comprises a processing module, an amplification module, and a control module. The processing module receives and processes an audio signal according to a control signal, and generates a processed signal. The amplification module coupled to the processing module, amplifies the processed signal to generate an output signal. The control module coupled to the processing module and the amplification module, receives the output signal to generate the control signal.

Abstract: Noise discrimination in signals from a plurality of sensors is conducted by enhancing the phase difference in the signals such that off-axis pick-up is suppressed while on-axis pick-up is enhanced. Alternatively, attenuation/expansion are applied to the signals in a phase difference dependent manner, consistent with suppression of off-axis pick-up and on-axis enhancement. Nulls between sensitivity lobes are widened, effectively narrowing the sensitivity lobes and improving directionality and noise discrimination.

Abstract: A sound reproduction device includes a filter, which detects a repeated bit sequence of a predetermined length in an input bit stream. When any repeated bit sequence having the predetermined length is detected, the input bit stream is applied to a filter for attenuating signals at the frequency corresponding to the predetermined length.

Abstract: A method and device for reducing error components of a multi-channel modulator. The method comprises the steps of inverting substantially half of the channels of a modulator, and reducing error components between said inverted channels and said non inverted channels by inductive and/or capacitive summing. The method is especially suitable for synchronized pulses and similar signals to be modulated.

Abstract: A system allows greater dynamic range in fixed-width sample representations of waveforms. Known properties of the waveform are used to determine an unlikely, or impossible, data condition to be used as an exception condition. Samples following the exception condition are assumed to be shifted by a predetermined amount so that their representation within the allowable dynamic range of the word results in an extended dynamic range. In a preferred embodiment, pre-processing of a waveform is used to create exception conditions and to shift portions of the waveform that would otherwise become clipped, so that the waveform peaks reside in a shifted position within the digital representation. Analogously, playback processing serves to detect the exception conditions and to restore the peaks of the waveform back to their original status for, e.g., audible digital-to-analog playback, high-resolution processing, etc.

Abstract: A noise canceller that a receiving signal received by a receiving antenna is input to, the noise canceller removing a pulse noise superimposed on the receiving signal to output the signal, the noise canceller comprising: a pulse noise detecting unit that detects the pulse noise superimposed on the receiving signal; a reception level detecting unit that detects the level of the receiving signal; a level hold unit that is triggered by the detection of the pulse noise with the pulse noise detecting unit to hold the level of the receiving signal in a process on the preceding stage of the output; and a level hold period setting unit that sets the level hold period when the level is held in the level hold unit in accordance with the level of the receiving signal detected by the reception level detecting unit.

Abstract: The invention describes a graphical method for detecting and adjusting audio overload conditions. The graphical user interface provides a user complete playback control of several audio tracks, detection of overload conditions such as audio clipping, and graphical methods to correct the overload conditions. The graphical interface provides drag handles which the user can use to adjust the various characteristics of an audio file. The characteristics, such as amplitude and temp, may be adjusted as a function of time.

Abstract: A method for improving a downlink signal received by a listener on a phone is disclosed. The method includes calculating an environment noise level of the listener and filtering and adjusting gain of the downlink signal based on the environment noise level.

Abstract: A voice signal clipping circuit comprising, an amplifier which amplifies an input voice signal to a permissible level and a clipper which clips an amplitude component in excess of the permissible level from an output signal from the amplifier.

Abstract: An output reducing system for a telephone system, e.g., implemented in a headset, includes an input line and an output line, wherein the input line transmits an incoming signal to an output transducer and the output line transmits an outgoing signal from an input transducer, and a compressor system is provided in the input line for reducing the output signal level.

Abstract: An acoustic signal level limiter provides a telephone handset/headset user protection against loud audible signals generated within a communications system. The acoustic signal limiter comprises an acoustic signal level attenuation circuit and at least one acoustic signal level relay circuit. Once activated, the acoustic signal level attenuation circuit creates an attenuation network that attenuates an electrical acoustic signal transmitted through the communications system. Acoustic signal level relay circuits are activated to further attenuate the electrical acoustical signal to prevent the acoustic signal level attenuation circuit from operating in a deep saturation mode and provide further hearing safety for the telephone headset/handset user. A fuse in series and “Zener Zap” shunting transistor diodes provide assured higher signal level protection.

Abstract: An apparatus for reducing noise component contained in an input signal has: a detecting device for detecting a level of the noise component; an adjusting device for adjusting a level of the input signal so as to make the level of the noise component equal to or lower than a predetermined threshold level; a reducing device for reducing a signal component of the adjusted input signal whose level is equal to and lower than the predetermined threshold level; and a restoring device for restoring a level of the adjusted input signal to the level of the input signal that has not been adjusted by the adjusting device yet.

Abstract: A noise reduction system is connected to at least two input channels for receiving input signals that have substantially identical program content but differing instantaneous noise transients. The system selects a filtered input signal that has the lowest instantaneous noise transient level. In another aspect, the noise reduction system removes instantaneous transient noise from an input signal by way of a blanker. The blanker includes a switch that opens whenever a band-pass filtered input signal is greater than the average level of the signal by a predetermined proportional amount. The input signal also includes continuous noise that is removed by a continuous noise suppressor. The continuous noise is removed by a low pass filter whose corner frequency is responsive to a differentiated input signal divided by the program component signal, or vice versa.

Abstract: The invention is a method, apparatus and computer readable program for eliminating overflow and underflow by re-adjusting or rectifying the audio levels of an audio signal in those portions where overflow/underflow occurs. In a first preferred embodiment a single stage scale down solution is provided in starting and end points of the overflow/underflow areas in the audio signal are detected, and a number of discrete sampling points along the audio signal curve between these two points are adjusted by a predetermined multiplier in order to reduce the positive level (in the case of overflow) or negative level (in the case of underflow) of each discrete sampling point. Using the single stage scale down solution of the present invention, a new audio signal is created having no overflow/underflow, yet maintaining the former generally smooth shape of the audio signal waveform and signal characteristics, without completely clipping the audio levels.

Abstract: An impulse noise reducer detects impulse noise in an audio signal by detecting and smoothing the high-frequency amplitude of the audio signal, attenuating the non-smoothed amplitude according to the smoothed amplitude, and comparing the attenuated amplitude with a threshold. Impulse noise is discriminated from high-frequency audio components because the latter tend to occur in longer-lasting bursts and are therefore attenuated more strongly. The impulse noise reducer is simplified because it does not have to perform intermediate-frequency signal processing, and its sensitivity is not affected by adjacent-channel signals because these signals are substantially absent from the audio signal. The impulse noise reducer can be implemented by digital signal processing, and is suitable for use in a medium-wave AM audio broadcast receiver.

Abstract: Clicks in voice and music signals are removed by detecting and interpolating the click. The use of a permanently set threshold value above the peak value of the audio signal is known for detecting the clicks, and interpolation carried out on the basis of the sample values before and after a click is known for filling them in. Disadvantages of these methods are, on the one hand, that many clicks do not have a larger amplitude than the peak signal and therefore remain undetected, and, on the other hand, interpolation is a highly complex procedure and is limited to a maximum click density. The invention provides for an error signal (en) to be determined by adaptive filtering (AF). A sample value is designated as distorted in the input signal (xn) if the absolute instantaneous value in the error signal (en) exceeds a dynamically adapted threshold value.

Abstract: A system and method for reducing noise in an audio signal. The system includes an encoder, a decoder and a single feedforward sidechain for generating a control signal that is used as input to a variable gain element in the encoder and in the decoder. The system is substantially audibly transparent, minimizes audible noise modulation, minimizes dynamic distortion of low frequency signals due to a level detector tracking these waveforms, minimizes distortion of signals due to phase shift in a transmission channel, minimizes distortion of fast, high-level transients due to the attack time of the level detector, and has virtually no effect on the audio signal if the level of the signal stays within a reasonable range around the nominal signal level.

Abstract: An ambient-aware headset is described. A mixer coupled to an external audio source receives an external audio signal. The mixer is also coupled to an ambient audio source to receive an ambient audio signal. The mixer mixes the external audio signal and the ambient audio signal according to a specified relationship. A speaker coupled to the mixer emits the external audio signal and the ambient audio signal into an ear canal of a user after the external audio signal and the ambient audio signal have been mixed by the mixer.

Abstract: The anti-pop circuit includes a unity gain buffer with an input coupled to the source of the reference voltage and an output coupled to the input of the amplifier to accelerate the charging of the input coupling capacitor of the amplifier at every turn-on. The capacitor-charging buffer is automatically disabled before the turning-on of the amplifier. The charging buffer may be enabled at start up by generating an impulse of a pre-established duration at the turn-on instant by a monostable circuit or by disabling it upon verifying the decaying to zero of the charging of current of the input coupling capacitor. The circuit eliminates the popping noise at the turn-on without an excessive delay of the turning-on of the amplifier.

Abstract: An audio peak limiter in which the lower sideband of the modulation distortion spectrum is canceled. The peak limiter is realized by multiplying the input signal by a control signal. The Hilbert transform of the control signal is multiplied by the Hilbert transform of the input signal and the outputs of the two multipliers are added. This cancels the lower sideband of the modulation process. Various filters can be added to eliminate aliasing in digital realizations, to band-limit the output signal, and to control peak overshoot.

Abstract: An output reducing system for a telephone system, e.g., implemented in a headset, includes an input line and an output line, wherein the input line transmits an incoming signal to an output transducer and the output line transmits an outgoing signal from an input transducer, and a compressor system is provided in the input line for reducing the output signal level.