Abstract: A compander for noise reduction of a FM signal is described, wherein a group delay (&tgr;) linked to the generation of the compressor gain (cc(t)) is equalised during generation of the multiplex signal (m(t)), and a group delay (&tgr;) linked to the generation of the expander gain (ce(t)) is equalised during generation of the sum signal (us(t)) and the expanded difference signal (ue(t)). Alternatively or additionally the compressor gain and/or the expander gain is controlled by an auxiliary signal on the basis of a combination of sum signal and difference signal of the FM signal. Such companders avoid overmodulation in the transmitter.

Abstract: A signal processing system for permitting a user to set certain system parameters includes at least one compander, at least one system volume control for setting system gain, and at least one transform engine responsive to the user-set parameter or parameters for controlling the operation of the compander and the setting of the system volume control. The transform engine may operate by using kneepoints, attack and release parameters, gain calculation coefficients, zero dB offset values, or other techniques or parameters. A central power estimator mixer may also be included for integration of a plurality of companders.

Abstract: An apparatus and method for compressing and expanding an information signal for use in noise reduction which includes an input terminal to receive the information signal, a first frequency crossover circuit coupled to the input terminal and configured to divide the information signal into a first and second frequency band. A second frequency crossover circuit coupled with the first crossover circuit configured to receive at least a portion of the information signal within the first frequency band and to further divide the information signal into a third and fourth frequency band. A third frequency crossover circuit coupled with the first crossover circuit configured to receive at least a portion of the information signal within the second frequency band and to divide the information signal into a fifth and sixth frequency band.

Abstract: A microphone expander attenuates background noise in a digitized microphone signal from a wireless telephone by using a loss function that exhibits hysteresis when the signal is passing through a transition range. This allows the microphone expander to apply loss more effectively because it allows for decreases in speech levels without attenuation. An attenuation level is determined using a first loss function when an averaged signal, derived from the digitized microphone signal using an algorithm that causes the averaged signal to clearly exhibit speech components in the digitized microphone signal, increases from below a lower noise threshold. However, if the averaged signal increases to above an upper speech threshold and then decreases to below that threshold, the attenuation level is determined using a different loss function, which delays introducing loss into the microphone signal as compared to the first loss function.

Abstract: A method is provided for operating a hearing aid as well as a hearing aid having at least one input transducer for picking up an input signal and converting it into an electrical signal, having an A/D converter for converting the electrical signal into a digital signal, having a signal processing unit for processing the digital signal and having an output transducer that the electrical input signal be compressed before the A/D conversion based on the criterion of a compression characteristic, and that the compressed digital signal or a signal proceeding from it can be expanded after the A/D conversion based on the criterion of an expansion characteristic that is at least approximately inverse relative to the compression characteristic.

Abstract: A device for decoding audio signals subjected to a noise-reduction encoding such as a Dolby-B encoding comprises a plurality of processing blocks (612, 616, 620, 624, 628, 632, 636, 640) for generating, starting from an input signal (610) containing the audio signal subjected to encoding superimposed on a noise component, an output signal (626) consisting of a replication of the audio signal with the noise component reduced. The aforesaid processing blocks are implemented in a digital form and comprise a sliding-band filtering structure (612) fed with the input signal (610) and designed to generate a filtered signal (614). The filtered signal (614) is fed, according to a general feedforward scheme, to an overshoot-suppression stage (616) in view of the generation, in an adder node (620), of a difference signal (622) starting from which the output signal is obtained via filtering (624).

Abstract: A method and system for selectively and variably attenuating audio data are disclosed. A high-volume control value as selected by a user is first received, and this high-volume control value defines a volume output level for high amplitude audio samples. An attenuation factor is then determined by utilizing the high-volume control value. Each sample from an incoming audio data stream is conditionally attenuated with the attenuation factor such that high amplitude audio data get compressed while low amplitude audio data remain unaffected. Finally, the attenuated samples are sent to an output.

Abstract: The present invention is a digital dynamic compression or automatic gain control (AGC) (10) adapted for use in high quality audio and hearing aids applications. An efficient digital AGC design employs two compact ROM-based tables (ROM_CSD, ROM_SPL) in addition to two comparators (COMP_A, COMP_B) and several registers (REG_A, REG_B, ADDR_A, ADDR_B). While one ROM stores the values of discrete input signal levels, the other contains gain codes based on a canonical signed digit (CSD) coding approach that leads to a very simple gain multiplier (20). In many cases an extremely compact table for gain values can be achieved by reusing a single small-size ROM that behaves like one that is several time larger. Two design examples are shown to expound the insights of the new digital AGC design. For the less-than-half-dB-gain-step cases only two adders are required for the multiplier whereas just three adders are needed in the situations with less than quarter-dB gain steps.

Abstract: A variable signal attenuating circuit which has a function of preventing dropout of an attenuated signal even if a large attenuation value is set. A dynamic range compression processing unit is provided in an input unit of the variable signal attenuating circuit for compressing the dynamic range of an input signal within a predetermined range in accordance with a set attenuation value, such that the signal passing through the compression processing unit is attenuated by the set attenuation value.

Abstract: Method and apparatus for generating a reference signal includes parsing one or more input signals into a plurality of split signals in accordance with a predetermined criteria, generating split reference signals for each of at least two of the split signals, and combining a plurality of the split reference signals into a single reference signal. Multiple reference signals may be generated by mutiple pluralities of split reference signals. The predetermined criteria may include bandsplit filtering, and the output signals may be used for equalization, among other applications.

Abstract: A method of matching input amplitudes in a system wherein one or more of a plurality of inputs may be selected, with each input capable of having different characteristics, involving selecting an input signal and mapping the input signal to a predetermined signal amplitude through the use of level matching logic. The level matching logic may include a gain cell for increasing or decreasing the amplitude of the input signal.

Abstract: A method of signal compression of a signal to reduce the large dynamic range, for example, of the output signal of a CD player in such a way that soft passages of a piece of music are reproduced in somewhat louder manner and very loud passages of a piece of music some are reproduced in somewhat less loud manner. The signal compression circuit used therefor can be integrated completely. The audio signal is fed over a volume control, which can be adjusted by the user, and to this end, a digital setting signal in accordance with the adjustment made on the user side, is supplied via a bus line. When the audio signal is strongly increasing, a tracking signal is formed of said signal, with the amplitude thereof being dependent on the medium amplitude of the audio signal. A window comparator compares the tracking signal with a reference value, as a consequence of which the contents of the up/down counter are reduced. This reduced value is added via an adder to the digital setting signal for the volume control.

Abstract: An apparatus and method for error concealment of compressed audio data are provided. The apparatus includes a packet-unit controller, a threshold controller, coupled to the packet-unit controller, a scale detector coupled to the threshold controller, and an error-correction circuit coupled to the packet-unit controller and the scale detector.

Abstract: Computer processor method and apparatus for creating an audio multiplier control signal for controlling the dynamic range of a recorded audio work. The technique includes determining an envelope of the amplitude of amplitude versus time values the audio signal, and then determining, for values of the envelope, respective minimum and maximum multiplication factors (MinMF and MaxMF) that can be multiplied times the values such that the products are above a predetermined minimum amplitude and below a predetermined maximum amplitude of the dynamic range. Then a control signal function of amplitude versus time is created such that all values of the control signal function at particular times are between respective MinMF and MaxMF values for the times, and such that segments of the control signal function have reduced slopes. Also disclosed is a method of creating a reduced-slope series of line segments passing through a pair of Max and Min limiting functions specifying y values with respect to a variable x.

Abstract: A dynamic range expander circuit is arranged to receive and process an input audio signal to provide an expanded output signal having a greater dynamic range than the input signal. The expander circuit may be structured with a rectifying multiplier module to receive an alternating current (AC) portion of the input audio signal and generate a direct current (DC) voltage that is applied to a gain control input of a dynamic amplifier module. The dynamic amplifier module also including an input to receive the input audio signal and an output providing the expanded output audio signal.

Abstract: Method and apparatus for generating a composite audio signal clocked to a selected frequency from left and audio input signals. A video/audio signal generator transmits the left and right audio input signals, together with a video signal having a horizontal synchronization signal embedded therein. If copy protection has not been applied to the video signal, the embedded horizontal synchronization signal will have a frequency of 15.734 KHz. If applied to the video signal, copy protection techniques will shift the frequency of the horizontal synchronization signal. To avoid distortion of the composite audio signal, a switch configured to selectively couple the video signal or a secondary video signal generated by an internal video signal generator to an audio signal processor which generates the composite audio output signal from the left and right audio input signals. Embedded within the secondary video signal is a horizontal synchronization signal having a frequency of 15.734 KHz.

Abstract: An acoustic conditioning apparatus. The mechanism includes an attenuator to limit excessive noise levels. The attenuator includes a circuit for compressing an output sound pressure level below a desired level, such as 90 dBA over an eight-hour period. The mechanism also includes an amplifier for conditioning ambient sound level in accordance with the normal needs of a hearing impaired person using the conditioning apparatus. Additionally included is a switch in an electrical circuit to enable selection of attenuation of excessive noise levels or normal amplification. The switch can be controlled either manually by means of a toggle on a housing enclosing the various components, or automatically in response to the sensed noise level.

Abstract: The Triple-Channel Slide Switch connects three audio channels into a single mixing device, allowing the user to access any of these channels through the single stroke manipulation of one switch.

Abstract: A digital audio decoder decodes or expands compressed data such as bit stream data, which are compressed based on the MPEG/Audio standard. Inverse quantization circuits perform inverse quantization on plural bit stream data, which are supplied thereto in connection with multiple channels respectively, thus producing inversely quantized data with respect to a prescribed number (e.g., thirty two) of sub-band samples respectively. The inversely quantized data are combined together among the multiple channels with respect to the prescribed number of the sub-band samples respectively. Then, a filter bank synthesizes together combined data corresponding to all of the sub-band samples, thus reproducing original digital audio signals. Multipliers are provided for use in gain control on the inversely quantized data with respect to the sub-band samples respectively.

Abstract: A method for controlling the peak levels of an FM composite signal is described. First, the FM signal is clipped to obtain an error signal. This signal comprises the “clippings” resulting from the clipping operation. Next, the peaks of the error signal are determined. Then, a half-cosine wave is fitted to each of the peaks. The resultant half-cosine-interpolated signal is filtered to provide a filtered error signal. This error signal is subtracted from the FM composite signal.

Abstract: An encoding apparatus is provided for encoding a wideband digital signal, including an input for receiving the wideband digital signal, a splitter unit for splitting the wideband digital signal into a number of M sub signals, a data reduction unit for data reducing the sub signals and a formatting unit for assembling digital information including the data reduced sub signals into a digital output signal having a format suitable for transmission or storage. During a repeated encoding and decoding of a wideband digital signal in a subband encoding/decoding system or a transform encoding/decoding system, signal degradation may occur. In order to avoid this, the apparatus further including a variable delay unit coupled between the input and the splitter for realizing a delay, the length of which is controlled by a control signal, and a control signal generator unit for generating the control signal.

Abstract: A small-sized audio signal reproducing apparatus for hearing reproduced audio signals with the aid of a headphone is disclosed. Digitized and compression encoded audio signals, stored in a semiconductor memory, are read out so as to undergo a decoding operation, which is an inverse operation to compression encoding, to reproduce the audio signals, and the reproduced signals are heard by the headphone. The apparatus may be significantly reduced in size and weight as compared to the apparatus in which a tape or a disk is used as the recording medium.

Abstract: Output control system and method includes calculating gain for an input signal, detecting a predetermined condition, and modifying the gain calculation in accordance with one or more kneepoints to provide a varying output signal. A companding ratio is then established in accordance with the kneepoints.

Abstract: An encoding device converts four-channel or five-channel audio signals to two-channel audio signals, which are then subjected to compression in accordance with the MPEG standard. Matrix coefficients are calculated based on the two-channel audio signals. A decoding device receives the compressed two-channel audio signals together with the matrix coefficient. The compressed two-channel audio signals are expanded and are then subjected to prescribed arithmetic operations using the matrix coefficients. Thus, the decoding device reproduces the original audio signals. The decoding device, which is normally actualized by a digital signal processor (DSP), is noticeably reduced in amounts of calculations and is simplified in circuit configuration because of the elimination of calculations of matrix coefficients therein.

Abstract: The present invention comprises methods and systems for a dynamic audio processor for processing an audio signal prior to encoding. The audio signal is pre-processed to provide analog-to-analog modification that creates a preferred analog format that is effectively and efficiently ready to be converted into a digital data stream by an encoder, such as an A-to-D converter or codec. One possible application includes, for example, a web streamer that provides digital data streams from a server. The inventive arrangements are preferably carried by functional stand-alone components, integral components of a PC or other computing platform, or carried as part of the encoder. Preferably, the pre-processing comprises receiving the audio signal, nominalizing the signal to provide a level input, and compressing and equalizing the signal for outputting thereof.

Abstract: A handheld remote controller for a televisor that automatically prefixes a single-digit program channel selection, typically between 2 and 9, with a leading zero in response to a prolonged single digit keybutton actuation exceeding about 3 seconds or by an intended ersatz key entry, such as a star “*” key, either preceding or following the channel digit key entry. A HUSH control provides audio volume reduction intermediate between mute and normal volume levels, obtainable by pressing a HUSH keybutton, or automatically hushed concurrent with a program channel change. Maintaining a televisor's necessitous channel 3 or 4 selection when used in conjunction with a cable box, VCR or other intermediate device is assured by concurrently sending a reiterative channel 3 (or channel 4) tuning command to the televisor whenever a program channel change command or other function signal is dispatched to the cable box, VCR or other television signal tuning apparatus.

Abstract: A method for limiting the dynamic response of audio signals in audio terminals, wherein an input signal x is treated such that a defined maximum limit value ymax for an output signal y from the terminal is not exceeded, and wherein a predeterminable constant amplification value v is set, where for v≦1 the input signal x with a nominal level is reproduced with maximum amplitude≦ymax without distortion as output signal y, and where for v=const>1 a distortion of the signals occurs. In a first step the input signal x is compressed in its dynamic response to an intermediate value y1<x, and in a following second step the output signal y=v * y1, amplified with the constant amplification value v, is formed.

Abstract: In a hearing amplification device adapted to receive a sound signal, the hearing amplification device having at least one channel configured to receive an input representative of said sound signal, the improvement comprising the channel being further configured to provide (1) linear gain for an input representative of a portion of the sound signal having a sound level less than a compression threshold, (2) rapid compressive gain for an input representative of a portion of the sound signal having a sound level greater than the compression threshold, wherein the rapid compressive gain is less than the linear gain, and (3) adaptive control of the compression threshold. Preferably the rapid compressive gain is instantaneous. Adaptive compression threshold control may be achieved in response to a user input and/or to sound signal changes. By adaptively controlling the compression threshold, performance of the device can be optimized to match its environment.

Abstract: Variable attack and release system and method are disclosed for dynamically modifying the various elements of the system including modifying compander response time to dynamically adjust for changing parameters such as environmental or input signal changes to maintain output signals within predetermined limits. The system and method include permitting accurate and quick response to noise sources having a duration which exceeds a predetermined threshold while at the same time ignoring transient or short duration environmental noise.

Abstract: An acoustic conditioning apparatus. The mechanism includes an attenuator to limit excessive noise levels. The attenuator includes a circuit for compressing an output sound pressure level below a desired level, such as 90 dBA over an eight-hour period. The mechanism also includes an amplifier for conditioning ambient sound level in accordance with the normal needs of a hearing impaired person using the conditioning apparatus. Additionally included is a switch in an electrical circuit to enable selection of attenuation of excessive noise levels or normal amplification. The switch can be controlled either manually by means of a toggle on a housing enclosing the various components, or automatically in response to the sensed noise level.

Abstract: A digital audio coding apparatus includes a part which converts a frame of digital audio data into a frequency domain; a part which divides the digital audio data into a plurality of bands; a part which calculates an allowed distortion level by using an absolute hearing threshold for each divided band and assigns coding bits; a change part which changes the absolute hearing threshold adaptively on the basis of intensity distribution of the digital audio data in the frequency domain.

Abstract: A digital clipper is highly oversampled to decrease aliasing and increase accuracy. The difference between the clipper's input and output is then downsampled and added to the delayed, unclipped signal at 1× sample rate to achieve clipping. Filters operating at 1× can be placed in series with the downsampled differentially-clipped signal to achieve overshoot compensation, bandlimiting of the clipped signal, and other goals.

Abstract: A method of signal compression of a signal to reduce the large dynamic range, for example, of the output signal of a CD player in such a way that soft passages of a piece of music are reproduced in somewhat louder manner and very loud passages of a piece of music some are reproduced in somewhat less loud manner. The signal compression circuit used therefor can be integrated completely. The audio signal is fed over a volume control, which can be adjusted by the user, and to this end, a digital setting signal in accordance with the adjustment made on the user side, is supplied via a bus line . When the audio signal is strongly increasing, a tracking signal is formed of said signal, with the amplitude thereof being dependent on the medium amplitude of the audio signal. A window comparator compares the tracking signal with a reference value, as a consequence of which the contents of the up/down counter are reduced. This reduced value is added via an adder to the digital setting signal for the volume control.

Abstract: A small-sized audio signal reproducing apparatus for hearing reproduced audio signals with the aid of a headphone is disclosed. Digitized and compression encoded audio signals, stored in a semiconductor memory, are read out so as to undergo a decoding operation, which is an inverse operation to compression encoding, to reproduce the audio signals, and the reproduced signals are heard by the headphone. The apparatus may be significantly reduced in size and weight as compared to the apparatus in which a tape or a disk is used as the recording medium.

Abstract: The present invention is a signal processing device which performs parallel processes A and B efficiently. There is a deviation in the throughputs of the process A and B in processing an audio signal. First to Nth sub signal processing sections have capabilities to complete the process A within a period (N×T). A main signal processing sections has a capability to complete the process B within a period T. Efficient signal processing can be achieved by processing an input digital signal by means of distinct sub signal processing devices one after another and then processing the signal by the main signal processing section. (N×T).

Abstract: Method and apparatus for generating a composite audio signal clocked to a selected frequency from left and audio input signals. A video/audio signal generator transmits the left and right audio input signals, together with a video signal having a horizontal synchronization signal embedded therein. If copy protection has not been applied to the video signal, the embedded horizontal synchronization signal will have a frequency of 15.734 KHz. If applied to the video signal, copy protection techniques will shift the frequency of the horizontal synchronization signal. To avoid distortion of the composite audio signal, a switch configured to selectively couple the video signal or a secondary video signal generated by an internal video signal generator to an audio signal processor which generates the composite audio output signal from the left and right audio input signals. Embedded within the secondary video signal is a horizontal synchronization signal having a frequency of 15.734 KHz.

Abstract: The present invention provides an audio enhancement apparatus and method which spectrally shapes harmonics of the low-frequency information in a pair of audio signals so that when reproduced by a loudspeaker, a listener perceives the loudspeaker as having more acoustic bandwidth than is actually provided by the loudspeaker. The perception of extra bandwidth is particularly pronounced at low frequencies, especially frequencies at which the loudspeaker system produces less acoustic output energy. In one embodiment, the invention also shifts signal from one audio signal to the other audio signal in order to obtain more bandwidth for the available loudspeaker to reduce clipping. In one embodiment, the invention also provides a combined signal path for spectral shaping of the desired harmonics and a feedforward signal path for each pair of audio signals.

Abstract: In a audio reproduction arrangement (2) such as a mobile phone terminal the decoded speech signal is compressed by compression means (14) in order to improve the understanding of the speech signal in the presence of strong background noise.

Abstract: The present invention is a method and apparatus for simply and inexpensively enhancing an electronic audio signal in such a way that the quality of audible sound produced from the audio signal more closely approaches that of the original sound heard live in an acoustically designed environment. The present invention restores the perception of harmonics that are normally missing in an electronic audio signal. The apparatus includes a circuit that causes an input audio signal to be distorted such that an enhanced audio signal is produced that exhibits an improved harmonic quality compared to that of the original input audio signal. This distortion is a non-linear amplification of enhancing harmonics or frequencies in the input audio signal. That is, a band of desirable harmonics or frequencies in the input audio signal are non-uniformly amplified across the band. This band of frequencies from the input audio signal can be a band of high frequencies and/or a band of low frequencies.

Abstract: The apparatus eliminates howls to permit a received sound to increase in sound volume to enable the apparatus to be used in high-noise areas, and comprises: a transceived sound volume control circuit 10 for controlling in sound volume a transceived sound, the circuit 10 being interposed between: an ear microphone 1 containing a microphone 4 and a speaker 5; and, a communication system 20 such as PHS. The circuit 10 comprises: a communication switching means 16 for performing a switching operation in communications; high-frequency-signal cutting means 17, 19, which attenuate a received sound signal in a transmission mode and then cut a high-frequency component of the signal; and, a transmitted-sound-signal attenuating means 15 for attenuating a transmitted-sound signal in a reception mode.

Abstract: A microphone output limiter which detects the amplitude of a microphone and produces a control signal which lowers the microphone's impedance when the amplitude of the audio signal exceeds a preselected threshold level. The output signal is detected at the output of the microphone or it may be detected at the output of a later pre-amplifier stage. Once a control signal is generated it is coupled to a variable impedance device which produces a shunt load impedance across the output of the microphone thereby lowering its impedance. The microphone, as well as the variable impedance device, may be coupled through impedance converters in order to facilitate component matching.

Abstract: A circuit is provided for rectifying and amplifying an AC input waveform to optimize the dynamic range of downstream circuitry, such as an analog-to-digital converter. The circuitry includes an inverting amplifier and a non-inverting amplifier. The inverting amplifier includes a selectable resistance network in a feedback loop that permits the gain to be adjusted by appropriate selection of conductive states of solid state switches. The non-inverting amplifier includes a selectable resistance network on an input line. A control circuit, such as a microprocessor, monitors the output of the A/D converter and controls the conductive state of switches in the feedback and input networks to maintain the digital output within a desired portion of the dynamic range of the A/D converter. Several discrete gains may be provided and programmed in accordance with a predetermined selection scheme.

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: The generation of an audio output sound from an electronic communications device includes monitoring a level of an electrical power supply for the electrical communications device, and adjusting an audio input signal responsive to changes of the level of the electrical power supply for the electronic communications device. The adjusted audio input signal is amplified, and an output sound is generated responsive to the adjusted and amplified audio input signal. In addition, an audio output signal can be generated responsive to received sound including an echo of the output sound, an estimate of the echo of the output sound can be generated using the adjusted audio input signal and a model of an echo path of the output sound, and the echo portion of the audio output signal can be reduced using this estimate.

Abstract: The present invention relates to a signal compressing circuit which amplifies an input signal on a logarithmic scale in an appropriate range, and has a variable gain stage, an input signal detector, and an automatic level controlling circuit, the automatic level controlling circuit including: a first comparator for comparing the input signal to a first reference voltage, and generating a signal proportional to the difference between them; a second comparator for comparing the input signal to a second reference voltage, and generating a signal proportional to the difference between them; and a current mirror for generating a controlling current in proportion to the output signal of the first and second comparators, thus controlling the gain of the variable gain stage in two steps so as to generate stable output signals.

Abstract: Again control arrangement for an auditory prosthesis, such as a hearing aid or cochlear prosthesis, is disclosed. The output of a controlled amplifier 11 is processed so as to derive an average noise floor level. This is compared to a predetermined level, and the gain adjusted slowly up or down in response to the comparison. Hardware and software implementations are disclosed.

Abstract: A voice replacing method and system that uses the volume of the karaoke user's voice to control the volume of the lead singer's original vocal outputted to the audience. The volume of an original lead singer's voice is determined by the volume or vocal-amplitude level of the karaoke user's voice. The vocal-amplitude of the karaoke user is sensed using a vocal sensor. A vocal-amplitude digital waveform is created corresponding to the vocal-amplitude. A smoothed exponential waveform is created byan amplitudefollower corresponding to the digital waveform. The volume of the original lead singer's voice is adjusted by a sliding balance according to the smoothed exponential waveform. This exponential waveform allows for a continuous and smooth volume adjustment of the original lead singer's voice according to changes in volume of the karaoke user's voice.

Abstract: The disclosed BTSC encoder includes a left high pass filter means for receiving a digital left channel audio signal and for digitally high pass filtering the digital left channel audio signal and thereby generating a digital left filtered signal; a right high pass filter means for receiving a digital right channel audio signal and for digitally high pass filtering the digital right channel audio signal and thereby generating a digital right filtered signal; a matrix means for receiving the digital left and digital right filtered signals, and including means for summing the digital left and digital right filtered signals and thereby generating a digital sum signal, and including means for subtracting one of the digital left and digital right filtered signals from the other of the digital left and digital right filtered signals and thereby generating a digital difference signal; a difference channel processing means for digitally processing the digital difference signal; and a sum channel processing means for d

Abstract: An automatic dynamic range control circuit is provided with a fixed-gain first circuit means and a variable-gain second circuit means having substantially the same propagation delay as the first circuit means. The automatic dynamic range control circuit holds the peak value of an input signal, selects the first circuit means when this peak value is smaller than a reference signal level, and selects the second circuit means when the peak value exceeds this reference signal level. When the held peak value is smaller than the reference signal level, the second circuit means is operated at the same gain relative to an input signal as the first circuit means by having its gain set by the reference signal. When the held peak value exceeds the reference signal, the gain of the second circuit means is limited by this held peak value and its output level is kept at or below a prescribed level.

Abstract: A method and an apparatus for restoration of dynamic range to electronically amplified acoustic signals. The apparatus includes in combination: (i) a first input for accepting electrical input signals; (ii) a first output for supplying the electrical input signals to a signal conditioning apparatus; (iii) a second input for accepting conditioned electrical input signals from the signal conditioning apparatus; (iv) a gain-controllable amplifier having a signal input coupled to said first input and an output coupled to a second output for providing dynamic range restored output signals; and (v) a first circuit, the first circuit for controlling gain of the gain-controllable amplifier such that the gain-controllable amplifier has maximum gain when the electrical input signal has a predetermined maximum value and such that the gain-controllable amplifier has minimum gain when the electrical input signal has a predetermined minimum value.