Abstract: An improved dynamics processing control system incorporates an integration release window to virtually eliminate ripple in the final control signal used to control a Voltage Controlled Amplifier in a signal processor. The input audio signal is level detected using a logarithmic level detector and filtered to provide a very fast time constant. The fast release time constant signal is clamped at a maximum level equal to the user defined threshold and filtered by a second filter providing a very long release time constant. The long release time constant is dynamically varied by the fast time constant to provide an adaptive slow time constant. The fast time constant signal modifies the adaptive slow time constant when the difference between the fast time constant and the slow time constant exceeds a predefined integration release window. The integration release window is based on a minimum number of decibels which is larger than the maximum possible ripple in the fast time constant signal.

Abstract: In an audio system with a plurality of input signals, the input signals are prioritized from least important to most important. The level of the aggregate of all of the input signals is compared to a threshold level. If the level is greater than the threshold, the levels of one or more of the input signals are reduced one at a time, in order from the least important input signal to the most important input signal.

Abstract: An audio device for improved talker discrimination is provided. To improve suppression of close talker interference, i.e., of a person talking in close proximity to the user of the audio device, the audio device comprises at least an audio input to receive a voice input signal; a filter bank, configured to provide a plurality of sub-band signals from the voice input signal; a multi-band audio expander, arranged to receive at least a group of the sub-band signals and configured to conduct audio expansion on the group of sub-band signals to provide gain-controlled sub-band signals; and an audio output, configured to provide a voice output signal from at least the gain-controlled sub-band signals.

Abstract: An audio processing apparatus includes a digital filter, a signal combiner, and a dynamic-range processor (DRP). The digital filter is configured to receive a filter-input audio signal and to filter the filter-input audio signal, so as to produce a filter-output audio signal. The signal combiner is configured to combine the filter-input audio signal with the filter-output audio signal, so as to produce a combined audio signal. The DRP is configured to apply to the filter-output audio signal a dynamic-range correction that depends non-linearly on the combined audio signal.

Abstract: An automatic gain control circuit controls a gain of a burst mode amplifier. A peak detector includes an input coupled to an output of the amplifier. A plurality of resistors is coupled in series between an input of the first amplifier and the output of the first amplifier for setting the gain of the amplifier. A first gain stage is responsive to an output signal of the peak detector for disabling a first resistor of the plurality of resistors to alter the gain of the first amplifier. A second gain stage is responsive to the output signal of the peak detector for disabling a second resistor of the plurality of resistors to alter the gain of the first amplifier. A comparator responsive to the output signal of the peak detector causes a pulse generator to enable the first gain stage and second gain stage each burst mode.

Abstract: Loudspeaker systems, which for technical reasons are not suitable for emitting strong bass signals, can use so-called virtual bass systems. Therein, low frequencies are replaced by their harmonics. However, virtual bass cannot always adequately replace real bass, such that tonal discrepancies may result. Methods and systems are disclosed to improve the bass reproduction of virtual bass by mixing the generated harmonics with a reduced original bass component of the input audio signal. The mixing ratio of this blend can be variable and can be determined automatically. For example, the mixing ratio can change when a level threshold is exceeded, when a temperature rises above/drops below a threshold, a calorimetric threshold is exceeded, or at fixed times of day.

Abstract: Example embodiments disclosed herein relate to signal processing. A method for decomposing a plurality of audio signals from at least two different channels is disclosed. The method comprises obtaining a set of components that are weakly correlated, the set of components generated based on the plurality of audio signals. The method comprises extracting a feature from the set of components, and determining a set of gains associated with the set of components at least in part based on the extracted feature, each of the gains indicating a proportion of a diffuse part in the associated component. The method further comprises decomposing the plurality of audio signals by applying the set of gains to the set of components. Corresponding system and computer program product are also disclosed.

Abstract: An audio processing method includes the following operation: dividing an audio file into a plurality of audio segments, in which a processing of a first audio segment of the audio segments includes the following operations: analyzing a first lowest energy value in a spectrum of the first audio segment; comparing the first minimum energy value with a preset energy value, and using a higher one as a first noise floor; generating a first processed audio segment according to the first noise floor and the first audio segment; compressing the first processed audio segment to produce a compressed audio segment; and sending the compressed audio segment to an audio playback device.

Abstract: The present invention provides methods and systems for digitally processing audio signals in two-channel audio systems and/or applications. In particular, the present invention includes a first filter structured to split a two-channel audio input signal into a low frequency signal and a higher frequency signal. An M/S splitter is then structured to split the higher frequency signal into a middle and a side signal. A detection module is then configured to create a detection signal from the middle signal, which is used in a compression module configured to modulate the side signal to create a gain-modulated side signal. A processing module is then structured to combine the low frequency signal, middle signal, and the gain-modulated side signal to form a final output signal.

Abstract: A system that performs automatic volume control and leveling (AVCL) based on audio category and desired volume level. The system may select different settings for audio data associated with different audio sources (e.g., content providers), audio categories (e.g., types of audio data, such as music, voice, etc.), genres, and/or the like. For example, the system may distinguish between music signals and voice signals (e.g., speech) and may apply a first gain curve for the music and a second gain curve for the speech. The gain curve may include an adaptive noise gate to reduce in-channel noise, such as by attenuating quiet sounds. The system may select the gain curve based on a desired volume level, and in some examples may modify the gain curve based on a signal-to-noise ratio (SNR) of the input audio and/or based on environmental noise.

Abstract: A handheld electronic, a sound produce system and control method of sound producing thereof are provided. The sound produce system includes a speaker, a feedback unit, a gain unit and a compensation unit. The speaker generates audible sound according to an output audio signal. The feedback unit generates a first detected parameter, wherein the first detected parameter represents a detected excursion. The gain unit receives an input audio signal, and amplifies the input audio signal to generate a gained audio signal according to a gain value. The compensation unit receives the gained audio signal and the first detected parameter, and predicts an excursion of the speaker corresponding to the gained audio signal to generate a predicted excursion parameter, and compensates the gained audio signal to generate the output audio signal according to the first detected parameter and the predicted excursion parameter.

Abstract: To control loudness during a junction between different types of broadcast content, such as a junction between program and commercial or promotional content, representative loudness values for content respectively before (P) and after (C) the junction are received from a playout automation system. A time-varying gain control is applied before and after the junction in order to smooth loudness around the junction. The audio gain is smoothly increased prior to the junction to a gain (P+C)/2P times higher than the original gain value. Then, the gain is reduced shortly before the junction to a value (P+C)/2C times lower than the original gain value. After the junction, the gain is returned smoothly to the original value.

Abstract: A method for automatically lowering the playing volume is disclosed. The method includes the following steps of: obtaining a first value according to a first playing volume and a first accumulated playing time of a sound player; determining whether the first value reaches a first threshold value; if yes, then lowering the first playing volume of the sound player to a second playing volume; obtaining a second value according to the second playing volume and a second accumulated playing time of the sound player; determining whether the second value reaches a second threshold value; if yes, then lowering the second playing volume of the sound player again, wherein the second threshold value is less than the first threshold value.

Abstract: A videoconferencing endpoint includes at least one processor a number of microphones and at least one camera. The endpoint can receive audio information and visual motion information during a teleconferencing session. The audio information includes one or more angles with respect to the microphone from a location of a teleconferencing session. The audio information is evaluated automatically to determine at least one candidate angle corresponding to a possible location of an active talker. The candidate angle can be analyzed further with respect to the motion information to determine whether the candidate angle correctly corresponds to person who is speaking during the teleconferencing session.

Abstract: The present invention provides methods and systems for digitally processing audio signals in two-channel audio systems and/or applications. In particular, the present invention includes a first filter structured to split a two-channel audio input signal into a low frequency signal and a higher frequency signal. A M/S splitter is then structured to split the higher frequency signal into a middle and a side signal. A detection module is then configured to create a detection signal from the middle signal, which is used in a compression module configured to modulate the side signal to create a gain-modulated side signal. A processing module is then structured to combine the low frequency signal, middle signal, and the gain-modulated side signal to form a final output signal.

Abstract: The present application describes signal enhancements to reduce bone conduction sensations of a wearable computing device and applications thereof. An example apparatus includes a wearable computing device comprising a bone conduction transducer (BCT) configured to receive and be driven by an audio signal, a processor, and a data storage comprising instructions executable by the processor to: (1) determine an input gain level of the audio signal at a frequency range; (2) compare the determined input gain level at the frequency range to a threshold gain level at the frequency range; (3) based on the comparison, apply a multi-band compressor (MBC) configured to process the audio signal to reduce gain in at least a portion of a mid-band frequency range of the audio signal; and (4) drive the BCT with the processed audio signal.

Abstract: The invention is directed to systems, methods and computer program products for adjusting audio properties after recording an audio. An exemplary method includes recording audio using an audio capturing system, wherein the recorded audio is associated with an audio level; determining one or more audio signal properties associated with the recorded audio; comparing the audio level with a first predetermined threshold level; and in response to determining the audio level is greater than a first predetermined threshold level, adjusting the recorded audio based at least partially on modifying the one or more determined audio signal properties, wherein the adjusting step is performed a finite time interval following the recording step.

Abstract: A system for processing audio data is disclosed that includes a plurality of gain adjustment devices, each gain adjustment device having an associated audio input frequency band. A plurality of control signal processing systems are configured to receive audio input data for one of the associated audio input frequency bands and to generate a gain adjustment device control signal. The gain adjustment device control signal is configured to decrease a gain setting of an associated gain adjustment device for a predetermined period of time as a function of a transient in the associated audio input frequency band.

Abstract: A multimode multiband wireless device includes a broadband RF power amplifier that receives RF signals and produces amplified RF signals in a cellular band in a broadband, and a coexist filter coupled to the input of the broadband RF power amplifier. The coexist filter can reject RF noise in a predetermined frequency range in the broadband adjacent to the cellular band.

Abstract: A limiter for an audio system prevents loud audio signals that exceed a threshold from being output. Output of the audio signals are delayed. When a loud signal exceeds the threshold, the gain applied to the delayed signals is gradually reduced so that by the time the loud signal reaches the output, the gain is at a level that reduces the loud audio signal to be within the threshold. Thereafter the gain is gradually restored to normal over a longer period of time than the audio signals are delayed.

Abstract: A monolithic Receiver Optical Sub-Assembly (ROSA) device is provided and a method for producing the device. The device comprises: at least one antenna configured to receive optical signals; at least one rectifier configured to rectify electrical signals being electrical representation of the received optical signals and having frequencies within an optical band range; and at least one amplifier, coupled to the rectifier and configured to amplify rectified electrical signals; and wherein the ROSA is also characterized in being a single monolithic device.

Abstract: A system and method applying Dynamic Range Control/Compression (DRC) to an audio signal. The dynamic range controller presented here differs from conventional DRC techniques by providing a much larger look-ahead time. In particular, the system and method takes advantage of the look-ahead by analyzing macroscopic loudness changes in the order of seconds as opposed to the microscopic changes most conventional DRCs are designed to control. This approach avoids most of the typical DRC distortions associated with conventional DRC techniques and preserves the micro-dynamics of the audio signal. Gain changes are applied at a rate comparable with manual volume adjustments by mixing and mastering engineers to balance a mix. Ideally, the DRC will approach what a professional sound engineer would do to reduce the dynamic range if there were only a volume control to accomplish the task on the final mix.

Abstract: A sound processing apparatus includes an information acquisition unit which acquires control information including at least one of mode information for designating a reproduction mode of the sound processing apparatus and attribute information for designating an attribute of an audio content represented by a sound signal, a frequency band expansion unit which performs frequency band expansion processing for adding an expanded component generated from the sound signal to the sound signal, and a control unit which changes parameters to be applied to the frequency band expansion processing in accordance with the control information acquired by the information acquisition unit.

Abstract: Systems and methods for compressing high-frequency signals are described in certain embodiments herein. According to certain embodiments, a high-frequency signal can be converted into a lower frequency signal so that it can be processed by one or more devices in a lower frequency infrastructure. In certain embodiments, the high-frequency signal can be compressed by certain signal conditioning components and an algorithm executed by a computer processor to at least receive a high-frequency signal, correct the high-frequency signal, determine a number of samples to be taken from the high-frequency signal (i.e., sample the high-frequency signal), store a value associated with the sampled signal, and generate a waveform that includes lower frequency content that may represent the original, high-frequency signal.

Abstract: A method and apparatus for fast, efficient, low noise power supply have been disclosed. In one implementation a switching power supply and a linear regulator provide efficiency and rapid response.

Abstract: A dynamic range compression circuit includes an attenuator that attenuates a signal at a predetermined node in an amplifier to reduce a gain of the amplifier and a gain controller that reduces the gain of the amplifier by the attenuator so that an amplitude of an output signal of the amplifier becomes an arbitrary output limit voltage in a case where an input signal having the same amplitude as that of an input-stage maximum voltage of the amplifier is input into the amplifier, and increases the gain of the amplifier by reducing a degree of attenuation of the attenuator according to a decrease of the amplitude of the input signal of the amplifier from the input-stage maximum voltage in a case where the amplitude of the input signal of the amplifier is smaller than the input-stage maximum voltage.

Abstract: The invention relates to a compressor and method for amplifying an input signal with a controlled gain. An output signal representing the input signal is amplified by an initial gain and a signal level of the input signal or of the output signal is compared with a threshold level. If the signal level is below the threshold level, the initial gain value is updated using an adaptive control characteristic, and if the signal level is above the threshold level, the initial gain value is updated using a fixed control characteristic or an adaptive control characteristic respectively. The adaptive control characteristic is dependent on the signal level and the fixed control characteristic is independent from the signal level.

Abstract: A system for limiting the excursion of an audio speaker. The system may include first lowpass filter circuitry configured to receive a system input signal and to generate a first signal. The system may further include clipping circuitry configured between the first lowpass filter circuitry and the first combiner circuitry. The system may include first highpass filter circuitry configured to receive the system input signal and to generate a second signal. The system may include second lowpass filter circuitry configured to receive a combined output signal from the first combiner circuitry. The system may include allpass filter circuitry configured to receive the system input signal and to generate a fourth signal. The system may include second highpass filter circuitry configured to receive the fourth signal from the allpass filter circuitry. The second combiner circuitry may be configured to generate an excursion limiting output signal.

Abstract: A dynamic range compression circuit includes an attenuator that attenuates a signal at a predetermined node in an amplifier to reduce a gain of the amplifier and a gain controller that reduces the gain of the amplifier by the attenuator so that an amplitude of an output signal of the amplifier becomes an arbitrary output limit voltage in a case where an input signal having the same amplitude as that of an input-stage maximum voltage of the amplifier is input into the amplifier, and increases the gain of the amplifier by reducing a degree of attenuation of the attenuator according to a decrease of the amplitude of the input signal of the amplifier from the input-stage maximum voltage in a case where the amplitude of the input signal of the amplifier is smaller than the input-stage maximum voltage.

Abstract: A wireless device configured to support a wireless networking protocol may utilize signal processing techniques that can mitigate the effects of jammer signals. For example, when a measured power associated with a digital sample of a received wireless signal is greater than a threshold, the wireless device may determine if the wireless signal corresponds to a wireless networking packet to be demodulated. If the wireless signal does not correspond to a wireless networking packet to be demodulated, the wireless device may adjust the threshold so that the power associated with the digital sample is less than the threshold. In other words, if the signal is a jammer signal, the wireless device may adjust its noise floor upward so that continuous reception of the same jammer signal does not trigger demodulation a second time.

Abstract: A method and/or apparatus for encoding and/or decoding an audio signal is disclosed, in which a downmix gain is applied to a downmix signal in an encoding apparatus which, in turn, transmits, to a decoding apparatus, a bit stream containing information as to the applied downmix gain. The decoding apparatus recovers the downmix signal, using the downmix gain information. A method and/or apparatus for encoding and/or decoding an audio signal is also disclosed, in which the encoding apparatus can apply an arbitrary downmix gain (ADG) to the downmix signal, and can transmit a bit stream containing information as to the applied ADG to the decoding apparatus. The decoding apparatus recovers the downmix signal, using the ADG information. A method and/or apparatus for encoding and/or decoding an audio signal is also disclosed, in which the method and/or apparatus can also vary the energy level of a specific channel, and can recover the varied energy level.

Abstract: Systems for an expander circuit are disclosed. An expander circuit comprising a transmit amplifier for amplifying electrical signals received from a microphone. A rectifier circuit is responsive to the electrical signals and generates a D.C. voltage. A dynamic resistor circuit is coupled to the transmit amplifier for regulating the gain of the transmit amplifier. The dynamic resistor circuit includes a bipolar junction transistor for receiving the D.C. voltage. The bipolar junction transistor is operated such that the collector current is zero and the bipolar junction transistor has a variable impedance seen by the transmit amplifier dependent upon the D.C. voltage.

Abstract: There are included an LPF (3) and an HPF (4) that are connected in parallel to the output of a pre-emphasis circuit (2). There is also included a gain adjusting circuit (6) that performs a gain adjustment of low-pass filter with respect to the frequency band to be passed through the HPF (4). The low frequency components of the frequency band of baseband signals outputted from the pre-emphasis circuit (2) pass through the LPF (3), while the high frequency components pass through the HPF (4). As to the outputs from the HPF (4), the gain of especially the higher part of the frequency band components to be passed through the HPF (4) is suppressed by the gain adjusting circuit (6), whereby the amplitudes of the baseband signals can be limited only for the high frequency range without using a limiter and further the peak values of the baseband signals can be inhibited from exceeding the maximum frequency deviation.

Abstract: A compound of formula (I) Wherein X, Y, ring A, ring B, l, m, R1, R2, R4 and R5 are as defined herein, to supress intracellular signal transduction or cell activation induced by endotoxin and to suppress cell responses due to the intracellular signal transduction and cell activation such as an excess generation of inflammatory mediators such as TNF-?, pharmacologically acceptable salts therefor, a preparation method therefor, and a medicament containing the aforementioned substituted cycloalkene derivative as an active ingredient which is superior in prophylaxis and/or treatment of diseases such as sepsis (septic shock, disseminated intravascular coagulation, multiple organ failure and the like), that are associated with intracellular signal transduction or cell activation induced by endotoxin and to cell responses to the intracellular signal transduction and cell activation.

Abstract: Musical distortion circuits are presented, typically for use with electric guitars, other electronic musical instruments or other sources of audio signals. In one embodiment, the musical distortion circuits of the present disclosure include a non-limiting clipping section. In one embodiment, the musical distortion circuits of the present disclosure include a waveshaping section. In some embodiments, the musical distortion circuits additionally include pre- or post-distortion or pre- or post-clipping sections.

Abstract: In a method and apparatus for transmission of signals, an input signal is supplied to at least one dynamic compressor and the dynamics of the signal is compressed by the dynamic compressor with a degree of compression. The compressed signal is subsequently adapted, in particular limited, in terms of its frequency bandwidth dependent on the degree of compression, such as directly dependent on the degree of compression or dependent on a reference value directly associated with the degree of compression, such as the input level of the signal before the dynamic compressor.

Abstract: Bandwidth expansion for audio signals by frequency band translations plus adaptive gains to create higher frequencies; use of a common channel for both stereo channels limits computational complexity. Adaptive cut-off frequency determination by power spectrum curve analysis, and bass expansion by both fundamental frequency illusion and equalization.

Abstract: A pulse train having a specific amplitude is transmitted from a transmitting circuit, and the amplitude at the receiving end is measured by a comparator. A coarse adjustment control unit determines a coarse adjustment value of the pre-emphasis intensity based on the measured amplitude. The pre-emphasis intensity is changed within the course pre-emphasis range, and the maximum receivable time width of a test signal is obtained by a reception judging unit 13 by using a fine adjustment stage.

Abstract: Compander architecture and method for ensuring that the components of an output signal are synchronized includes calculating a gain calculation for an input signal, detecting a predetermined condition such as a zero crossing or absence of a zero crossing within a specified period, and ensuring that the result of the gain calculation is providing to the system synchronously with the input signal. The input signal may be divided into one or more signals for processing, and the gain calculation may include one or more power estimations and one or more gain cells associated with the multiple signals, and may further include variable attack and release.

Abstract: A digital audio signal to be replayed is processed in a waveform thereof. A frequency bandwidth of the audio signal is expanded through conversion of a sampling frequency, and then the audio signal is low-pass-filtered with a low-pass cut-off frequency corresponding to the converted sampling frequency. An interval of time between two waveform peaks of the audio signal is detected, and then difference data between current data of the audio signal and past data thereof is calculated. The difference data are subject to weighting depending on the interval, and then output data are produced based on both the low-pass-filtered audio signal and the weighted difference data. This processing, which can be realized by activation of software, improves audio quality when compressed audio data is replayed.

Abstract: A method and an apparatus for detecting signals in an electromagnetically noisy environment by detecting electromagnetic signals comprising a noise signal and a known signal, and compressing the electromagnetic signal to thereby generate a compressed signal. The compressed signal is filtered to generate a filtered signal comprising substantially the known signal, and the filtered signal is expanded.

Abstract: A method for computing the “clippings” of an audio signal in the digital domain to prevent aliasing is disclosed. An offset signal is found by subtracting a threshold from samples of the audio signal. These are multiplied by a pulse after the pulse is lined up in time with the crossing of the audio signal at the threshold. The pulse is the sum of two step functions, both of which are bandwidth-limited.

Abstract: An audio signal processing circuit reduces noise in an incoming audio signal and is particularly useful in telephone communication systems utilizing one or more hands-free microphones. A preferred embodiment of the audio signal processing circuit includes a pre-emphasis circuit receiving the audio signal from a microphone or other transducer, an amplifier circuit receiving the pre-emphasized audio signal from the pre-emphasis circuit and a de-emphasis circuit receiving the amplified signal from the amplifier circuit. An output of the de-emphasis circuit provides the processed audio signal having an improved signal to noise ratio with minimum audible distortion. A preferred embodiment of the amplification circuit includes an amplifier defining a non-linear transfer function therethrough which provides low gain to the lower amplitude noise signals and higher gain to the higher amplitude audio signals.

Abstract: A multi-channel signal processing system adapted to provide binaural compressing of tonal inputs is provided. Such a system can be used, for example, in a binaural hearing aid system to provide the dynamic-range binaural compression of the tonal inputs. The multi-channel signal processing system is essentially a system with two signal channels connected by a control link between the two signal channels, thereby allowing the binaural hearing aid system to model behaviors, such as crossed olivocochlear bundle (COCB) effects, of the human auditory system that includes a neural link between the left and right ears. The multi-channel signal processing system comprises first and second channel compressing units respectively located in first and second signal channels of the multi-channel signal processing system. The first and second channel compressing units receive first and second channel input signals, respectively, to generate first and second channel compressed outputs.

Abstract: An audio codec and a method of compressing audio data makes use of a filterbank which automatically adapts itself to changes in the sampling frequency/bit rate to mimic the characteristics of the human auditory system. The algorithm used compares the bandwidth of each sub-band at a given depth with the critical bandwidth. If the critical bandwidth is less than the bandwidth of the sub-band, then the sub-band is split into two at the next level, and the process is repeated until the bandwidth of every sub-band is less than the critical bandwidth at the corresponding frequency. The codec thus automatically adapts itself to changes in sampling frequency/bit rate, which is particularly advantageous when very low bandwidths are in use.

Abstract: An audio system limits gain in individual audio channels having a shared clip detect signal from a multi-channel power amplifier such that only channels likely to be exceeding the distortion threshold are gain limited. In addition, any arbitrary channels may be grouped together for common gain limiting. The invention monitors the power level of each audio channel, compares the power level to a power threshold that indicates whether a high power condition with the potential to cause excess distortion exists or not, and makes a decision whether to activate each channel's gain limiter based on whether the corresponding clip detect signal is active and the high power condition exists simultaneously for that channel.

Abstract: A processing system for a charge coupled device (CCD) or CMOS imaging system includes a multi-mode, multiple current level, correlated double sample and variable gain (CDS/VGA) circuit for receiving data from a CCD system, subject to horizontal and vertical timing signals for the system which are locally generated by the processing system itself. The processing system particularly includes programmable timing circuitry for controlling the detection of pixel intensity values from elements of a two-dimensional pixel array, with a programable low-frequency master vertical timing circuit driving a high-frequency horizontal timing circuit, wherein the vertical and horizontal timing signals are independently locally provided to the array from the analog processor actually sampling the array. The architecture of the processing system further includes a correlated double sampler, a black level clamp, and an A/D conversion module.

Abstract: In a dynamics compressor, an analog signal to be compressed is supplied to a first amplifier stage as an input signal, and the output signal thereof is supplied to the next amplifier stage as an input signal, and so on for a number of successive amplifier stages. Each amplifier stage amplifies its input signal with a stage amplification until it reaches a stage limit level. The analog signal and the respective output signals of the amplifier stages are summed in a summation stage to form a sum signal. Above a minimal level, the magnitude of the curve of the sum signalroughly corresponds to an exponent characteristic of the magnitude of the analog signal.

Abstract: Apparatus and a concommitant method for generating an approximation of a logarithmically attenuated value of a digital signal in predefined increments. Specifically, the apparatus comprises a shift register, a latch and an adder. The shift register selectively shifts the digital signal a first number of bits to the right to produce a first shifted signal and then a second number of bits to the right to produce a second shifted signal. The latch is connected to the shift register and latches the first shifted digital signal while the digital signal is being shifted to form the second shifted signal. The adder is connected to the shift register and the latch and adds the first shifted signal to the second shifted signal to produce the attenuated value of the input digital signal.

Abstract: A method and system for monitoring, maintaining and analyzing statistical performance information in a signal processing system includes collecting data representing a plurality of aspects of the signal processing system, developing a histogram representative of the data in accordance with a chosen algorithm, and analyzing historical histogram data to generate condition signals for use by the signal processing system.