Abstract: Methods and apparatus to determine a level of inherent jitter for signals from a transmitter and a receiver, and modulate information onto a signal transmitted by the transmitter by using spot jitter (with bandwidth and center frequency modulation) and/or pulse width jitter in a region outside of a data region with inherent jitter to carry communication between systems.

Abstract: Methods and apparatus to determine a level of inherent jitter for signals from a transmitter and a receiver, and modulate information onto a signal transmitted by the transmitter by using spot jitter (with bandwidth and center frequency modulation) and/or pulse width jitter in a region outside of a data region with inherent jitter to carry communication between systems.

Abstract: A sound data transmitting apparatus that transmits digital sound data using first and second transmission channels for transmitting transmission signals of first and second forms respectively guaranteed and not guaranteed to obtain, as reception data, data same as transmission data, includes: a data separating unit that separates transmission digital sound data into first and second sound data that should be transmitted through the first and second transmission channels, respectively; first and second transmission processing units that process the first and second sound data separated by the data separating unit to obtain the transmission signals of the first and second forms, respectively; and a data transmitting unit that transmits, through the first and second transmission channels, the transmission signals of the first and second forms obtained by the first and second transmission processing units, respectively.

Abstract: In an audio network system constructed from a main node and a plurality of satellite nodes each having a plurality of ports, the main node generates and transmits a main packet including audio signals of a plurality of channels. Each satellite node selects one of the plurality of ports in turn, and confirms whether the main packet arrives at the selected port every predetermined period or not. When the main packet arrives at the selected port every predetermined period, the main packet is received via the port by continuing selection of the port. In the case where reception of the main packet is lost, one is selected from the plurality of ports in turn, the operation to confirm arrival of the main packet is restarted, and another port at which the main packet arrives at the present stage is automatically found.

Abstract: Method, apparatus and computer program product for processing an input stereophonic audio signal to thereby generate a converted stereophonic audio signal representing the input stereophonic audio signal, the input stereophonic audio signal comprising a left input audio signal and a right input audio signal, and the converted stereophonic audio signal comprising a first converted audio signal and a second converted audio signal. The first converted audio signal is generated based on the sum of the left input audio signal and the right input audio signal. The second converted audio signal is generated based on the difference between a first function of the left input audio signal and a second function of the right input audio signal. The first and second functions are adjustable to thereby adjust at least one characteristic of the converted stereophonic audio signal.

Abstract: A wireless audio device may realize power savings when processing multi-channel data, e.g. 2-channel stereo audio data, by splitting the multi-channel block data into separate data blocks for each channel, e.g. Left-channel data block and Right-channel data block, and processing each separate data block independently in the source device to generate respective data packets targeting corresponding sink devices, e.g. a Left earphone and a Right earphone. The source device may then transmit each data packet to a different corresponding sink device, e.g. a Left-channel packet to a Left earphone and a Right-channel packet to a Right earphone. The data packets may also include header information indicative of whether a next packet is intended for a given sink device, enabling sink devices to enter sleep mode when the header information indicates that a next packet is not intended for the given sink device, to save power.

Abstract: Audio encoding method and device comprising the transmission, in addition to the data representing a frequency-limited signal, of information relating to a temporal filter that can be applied to the entire broadened signal, both in its transmitted low-frequency part and in its reconstituted high-frequency part. The application of this filter allowing the reshaping the reconstituted high-frequency part and the correction of compression artifacts present in the transmitted low-frequency part. In this way, the application of the temporal filter, simple and inexpensive, to all or part of the reconstituted signal, makes it possible to obtain a signal of good perceived quality.

Abstract: A sound image localization apparatus comprises an L direct output section that produces an output signal by inputting an audio signal of a rear left audio input channel to a filter having a characteristic obtained by dividing RLD by LD, an L cross output section that produces an output signal by inputting the audio signal of the rear left audio input channel to a filter having a characteristic obtained by dividing RLC by LC, an R cross output section that produces an output signal by inputting an audio signal of a rear right audio input channel to a filter having a characteristic obtained by dividing RRC by RC, an R direct output section that produces an output signal by inputting the audio signal of the rear right audio input channel to a filter having a characteristic obtained by dividing RRD by RD, a first adding section that adds a difference signal between the output signal of the L direct output section and the output signal of the R cross output section to an audio signal of a front left audio input ch

Abstract: The invention provides a noise removal device and method capable of more proper interpolation on an input signal. The noise removal device is for removing noise in an input signal and includes: a noise detector detecting noise in an IF signal and outputting a noise detection signal; an interpolation controller determining a period and amount of interpolation for noise correction processing, based on the IF signal and the noise detection signal; and a noise gate processor performing the noise correction processing on the IF signal, based on the interpolation period and amount supplied from the interpolation controller. The interpolation controller sets a predetermined first interpolation period, based on a first noise detection signal inputted from the noise detector, and redefines a second interpolation period longer than the first interpolation period when a second noise detection signal is detected within the first interpolation period.

Abstract: A technique for reducing inherent noise in a real time digital audio wireless system. The technique allows for low power stereo analog to digital and digital to analog converters to be used in a battery operated real time digital audio wireless applications. The audio signal is sent to the left and right channels of an analog to digital converter with a preceding audio amp of different gain at the input to each channel. The signals are simultaneously digitized then combined to generate a new digital value resulting in noise reduction at lower amplitudes. This new data word is then transmitted via radio waves to a digital audio receiver. Once the data is received it is output simultaneously to a stereo digital to analog converter. The digital to analog conversion of both left and right channels is then summed using a resistor network and an audio amplifier. The audio amplifier output is then reduced to create further reduction of noise at all amplitudes.

Abstract: A demodulator (10) converts television signals to video baseband signals and audio baseband signals including stereo signals representing a right channel signal value and a left channel signal value. A DSP (60) recursively finds a preferred coefficient value for a scaling that reduces stereo separation due to amplitude variation of the right and left channel signal values. The preferred coefficient value is thereafter used for scaling the right and left channel signal values.

Abstract: A stereo decoder and a method therefor are provided. The stereo decoder receives a MPX signal from an FM demodulator, and comprises a first auto-calibration circuit, a band-pass filter, a second auto-calibration circuit, a slicer and a PLL circuit. The first auto-calibration circuit generates a first control signal. The band-pass filter generates the pilot signal by filtering the MPX signal with a center frequency set by the first control signal. The second auto-calibration circuit generates a second control signal. The slicer converts the pilot signal into a square wave signal. The PLL circuit comprises a voltage controlled oscillator for generating an oscillation frequency in response to the second control signal. The PLL circuit receives the square wave signal to generate the reference signal around the predetermined frequency in response to the oscillation frequency.

Abstract: According to one embodiment, generating images according to intersection points includes obtaining samples of signals from transmitter-receiver pairs. A transmitter-receiver pair is configured to transmit a signal and receive the signal reflected by an object. Intersection points are determined, where an intersection point indicates an intersection for integer multiples of a sample-time distance. A subset of samples corresponding to the intersection points is selected. Image data is generated from the selected subset of samples, where the image data is used to generate an image of the object.

Abstract: A parametric representation of a multi-channel audio signal describes the spatial properties of the audio signal well with compact side information when a coherence information, describing the coherence between a first and a second channel, is derived within a hierarchical encoding process only for channel pairs including a first channel having only information of a left side with respect to a listening position and including a second channel having only information from a right side with respect to a listening position. As within the hierarchical process the multiple audio channels of the audio signal are downmixed iteratively into monophonic channels, one can pick the relevant parameters from an encoding step involving only channel pairs carrying the information needed to describe the spatial properties of the multi-channel audio signal.

Abstract: A portable communication apparatus and method includes a portable AM transmitter assembly and antenna operably connected for broadcasting an AM signal to standard automobile radio receiver, such as on a travel information station frequency of 1610 KHz. The system is particularly constructed for mobility, durability, simplicity and ease of use, making it well adapted for use in emergency situations as well as for the broadcast of general community or travelers information at fixed or temporary locations during nonemergency times. For example, the entire present assembly 20 can be constructed with a total weight of less than about 130 lbs, allowing it to be easily moved by individuals and by low-weight-capable small capacity vehicles such as planes or boats or pick-up trucks.

Abstract: Disclosed is a stereo signal processing apparatus, in particular for a digital BTSC television decoder, comprising a sub-channel signal processing section which comprises an input for inputting an input sub-channel signal, a DBX expanding means and an output for outputting an output sub-channel signal. The particularity of the present invention is that said sub-channel signal processing section further comprises a phase error compensating means for correcting a phase error of said DBX expanding means so that at said output of said sub-channel signal processing section the phase of the output sub-channel signal is essentially constant or zero over a predetermined frequency range.

Abstract: A novel system and method for correcting the residual phase offset between a recovered pilot signal and the received stereo signal. The invention uses a Costas loop as an auxiliary loop in addition to the pilot recovery phase locked loop (PLL) to lock onto the stereo component itself. This auxiliary loop functions to generate a pilot to stereo component phase correction signal that is added to the stereo carrier phase The resultant phase is used to generate the recovered pilot carrier used to demodulate the stereo MPX signal. The Costas loop is activated together with the main pilot recovery PLL that locks onto the pilot tone in the demodulated MPX signal. The auxiliary Costas loop is operative to track and determine a residual phase error of up to several degrees.

Abstract: An apparatus and a method of automatically compensating a difference between audio signals of an analog broadcast and a digital broadcast depending on a difference of a modulation degree of an analog audio signal and a digital audio signal.

Abstract: In a frequency modulator, a VCO oscillates at a frequency according to a voltage applied to an input terminal. A divider divides an output signal of the VCO. A phase comparator compares the output of the divider with a reference clock signal and outputs a voltage corresponding to a phase difference. A loop filter is provided on a path leading from an output terminal of the phase comparator to the input terminal of the VCO, and the loop filter removes a high-frequency component of an output voltage of the phase comparator. A terminal for inputting a modulation signal is provided in the loop filter, separately from a terminal connected with the path leading from the output terminal of the phase comparator to the input terminal of the VCO.

Abstract: A method of encoding a multichannel signal, such as a stereophonic audio signal, including at least first and second signal components includes transforming at least the first and second signal components by a predetermined transformation into a principal signal including most of the signal energy and at least one residual signal including less energy than the principal signal. The predetermined transformation is parameterized by at least one transformation parameter. The method further includes representing the multichannel signal at least by the principal signal and the transformation parameter.

Abstract: A method of encoding a multi-channel signal having first and second signal components includes determining a set of filter parameters a prediction filter such that the prediction filter provides an estimate of the second signal component when receiving the first signal component as an input. The multi-channel signal is represented as the first signal component and the set of filter parameters. A corresponding decoding method and arrangements for encoding and decoding multi-channel signals are also provided.

Abstract: A stereo modulator converts an inputted audio signal to a stereo composite signal. A frequency modulator, which includes a PLL circuit, performs frequency modulation with the stereo composite signal outputted from the stereo modulator as being a modulation signal. A first programmable divider and a second programmable divider divide an external clock signal inputted from the outside by a first division ratio and a second division ratio set respectively and outputs. A first clock signal outputted from the first programmable divider is used as a reference clock signal for generating the stereo composite signal, and a second clock signal outputted from the second programmable divider is used as a reference clock signal of the PLL circuit.

Abstract: To retrieve RDS information by filtering and transforming an incoming multiplex signal (m(t)) into a RDS baseband signal, an amplitude modulated RDS signal (ma(t), mc(t)) is derived on basis of an intermediate signal (ma(t)) obtained during an extraction of a stereo-difference signal (md(t)) from the incoming multiplex signal (m(t)).

Abstract: A multiple signal carrier transmission apparatus and method for broadcasting stereo sounds to an audio output device uses three related carrier frequencies on which are modulated the sum of first and second stereo channel signals and the first and second stereo channel signals themselves. The composite multiple carrier frequency signal is transmitted to a receiver which separates the three carrier frequency signals to generate a separate left channel signal, a right channel signal and a composite center channel, monophonic, signal which is the sum of the first and second channel signals. A detector monitors the signal strength of each separate left and right channel signal to detect a decrease in either channel output below a threshold. When a decrease below the threshold is detected, the center sum signal replaces the left and/or right channel signal having the diminished signal strength for output through the appropriate speaker to minimize channel fade out.

Abstract: The present disclosure provides a system and method for transporting audio signals over conventional power lines within a building.

Abstract: A method and system is disclosed for digitally demodulating an analog signal. In one example, the analog signal contains a L+R signal, a L−R signal centered around a carrier signal, and a pilot signal. An exemplary system for digitally demodulating such a signal includes an analog to digital converter, three digital filters, clock reconstitution circuitry, L−R signal recovery circuitry, and channel recovery circuitry. The analog to digital converter converts the analog signal into a digital signal. The three filters separate the L+R signal, the pilot signal, and the L−R signal from the digital signal. The clock reconstitution circuitry reconstitutes a clock signal from the separated pilot signal. The L−R signal recovery circuitry recovers the separated L−R signal using the reconstituted clock signal. The channel recovery circuitry recovers a left channel signal and a right channel signal from the separated L+R signal and the recovered L−R signal.

Abstract: An improved speaker having a modified front plate with air intake apertures, a vented frame, and a voice coil supported by a diaphragm constructed of gas impermeable material connected at one end to the frame and at the other end to the voice coil. Operation of the voice coil causes the diaphragm to draw air through the air intake apertures where it is drawn past the voice coil and vented out through the frame. The diaphragm operates as an air pump forcing the air past the voice coil for supplying of cool are and exhausting of heated air. A unidirectional flap may be used on the vented frame to an create air flow current past the voice coil for higher efficiency. The improved speaker provides longevity to the voice coil, the voice coil to diaphragm coupling, and to the diaphragm itself.

Abstract: A system and method are disclosed for performing digital multi-channel decoding of a BTSC composite audio signal. Each subsequent stage of the digital multi-channel decoding process is performed at the lowest sampling rate that yields acceptable performance for that stage. Analog-to-digital conversion of the composite audio signal is performed first to generate a composite digital audio signal. After analog-to-digital conversion, all signal processing may be performed in the digital domain. The composite digital audio signal is digitally filtered to frequency compensate for variations caused by previous stages of processing, including IF demodulation. Digital channel demodulation and filtering are performed to isolate single channels of the composite digital audio signal such as SAP, L−R, and L+R channels.

Abstract: The invention addresses itself to the task of providing a medium-frequency stereo broadcast receiving circuit that does not require any alteration of the radio wave format used in the medium-frequency stereo broadcast system that has become the de facto standard, and that receives transmitted broadcast waves and removes, in the course of the demodulation process, disturbance that has affected the signal during its propagation, thereby improving the audio quality of the demodulated signal and obtaining the full potential of the stereo effect. To achieve this, the received medium-frequency stereo broadcast wave is converted to a single-sideband signal, and the sum signal (L+R) is demodulated from the phase term of this converted single-sideband signal. The difference signal (L−R) is demodulated from the phase term of the received medium-frequency stereo broadcast wave and from the demodulated sum signal output.

Abstract: The invention is a method for monitoring an environment by evaluating an infrasonic signal obtained from said environment. The method includes averaging said infrasonic signal (Vinfra) to provide an averaged infrasonic signal (Vav), mapping said averaged infrasonic signal (Vav) according to a function to provide an infrasonic noise signal (Vnoise); averaging said infrasonic noise signal (Vnoise) to provide an averaged infrasonic noise signal (Vav noise); offsetting said averaged infrasonic noise signal (Vav noise) with an offset value to provide an infrasonic level signal (Vlimit); comparing said infrasonic level signal (Vlimit) with said averaged infrasonic signal (Vav); and generating a trigger signal (30) if said averaged infrasonic signal (Vav) is greater than said infrasonic level signal (Vlimit).

Abstract: A background-sound control apparatus, a background-sound control method, readable recording medium containing a background-sound control program, and a video game apparatus improve effectiveness of background-sound data provided over a predetermined number of channels. When background sounds are produced through a speaker, a sound-level controller controls output ratios for sound levels defined in background-sound data on a channel basis and uses a sound-level control function to perform crossfading sound-level control.

Abstract: The invention relates to a communication receiver provided with spatialization means for generating a pseudo-stereophonic output signal from a monophonic input signal. The spatialization means comprise spectrum extension means for generating on a first output channel an output signal containing synthesis frequencies extrapolated from part of the original band of the monophonic input signal, while the input signal is sent over a second output channel.

Abstract: A method to denoise a stereo signal which comprises a stereo sum signal (Us(z)) and a stereo difference signal (Ud(z)), wherein a stereo to mono blending on basis of an attenuation of the stereo difference signal (Ud(z)) is performed, determines an attenuation of the stereo difference signal (Ud(z)) under consideration of the difference signal amplitude. Therefore, a stereo signal noise reducer which performs a stereo to mono blending on basis of an attenuation of the stereo difference signal, comprises an attenuation factor determination unit which determines an attenuation factor (Hatt) under consideration of the difference signal amplitude.

Abstract: A multiple signal carrier transmission apparatus and method for broadcasting stereo sounds to an audio output device uses three related carrier frequencies on which are modulated the sum of first and second stereo channel signals and the first and second stereo channel signals themselves. The composite multiple carrier frequency signal is transmitted to a receiver which separates the three carrier frequency signals to generate a separate left channel signal, a right channel signal and a composite center channel, monophonic, signal which is the sum of the first and second channel signals. A detector monitors the signal strength of each separate left and right channel signal to detect a decrease in either channel output below a threshold. When a decrease below the threshold is detected, the center sum signal replaces the left and/or right channel signal having the diminished signal strength for output through the appropriate speaker to minimize channel fade out.

Abstract: Audio signals are reproduced with consistent perception of total harmonic distortion even with varying spectral content of a particular audio signal without separately limiting different frequency bands within the audio signal. A distortion threshold representative of an amount of distortion to be allowed to be introduced by a power amplifier is determined in response to a frequency content of the audio signal. A distortion trigger compares a measured average distortion signal and the distortion threshold and reduces the controllable gain of the amplifier if the average distortion signal exceeds the distortion threshold.

Abstract: An apparatus and method is provided for transmitting “masked” stereo audio signals in a signal distribution network such as a cable television system within the bandwidth constraints of conventional frequency allocation schemes. In various embodiments, left and right stereo audio signal components (L+R and L−R) are used to amplitude modulate a carrier using independent sideband (ISB) modulation without the transmission of a pilot signal. A jamming signal or low bandwidth data signal may be transmitted in place of the normal pilot tone. Various aspects of the invention contemplate the use of Hilbert transforms to perform the ISB modulation and demodulation.

Abstract: An amplitude demodulator for demodulating a quadrature-amplitude modulated stereo signal for radio receivers, where a digital intermediate-frequency signal is constructed from a received signal, the intermediate-frequency signal is converted into a baseband, creating two orthogonal components thereby, a magnitude signal and a phase signal are derived from the orthogonal components, a tangent is constructed from the phase signal, and the tangent is multiplied by the magnitude signal to create a stereo difference signal.

Abstract: A method and apparatus for reducing peak-to-average power ratio in a paging transmitter broadcasting an independent sideband amplitude modulation protocol is disclosed. A digital signal, for example a digital voice signal, is provided to a control gain generator which generates a digital gain sequence. The digital gain sequence is such that during peak values in the digital signal the gain is low and during low values in the digital signal the gain is high. The digital gain sequence is multiplied with the digital signal to compress the dynamic range of the digital signal.

Abstract: A modulation scheme (600) useful in a voice paging system in which both of two orthogonal modulation components (500 and 510) are used to carry two halves of a single voice message destined for a receiver, or two separate voice messages for a receiver. The single voice message is transmitted in half the time.

Abstract: A controller for an electrical switching device such as a motor starting contactor including separable contacts, an operating mechanism for operating the separable contacts and a control mechanism responsive to a plurality of control sources for controlling the operating mechanism, includes a plurality of light emitting diode indicators each of which corresponds to one of the control sources; a membrane pushbutton; and a selection circuit, responsive to the pushbutton for selecting one of the control sources, including a circuit for detecting actuation of the pushbutton; a circuit responding to the actuation for sequentially indicating representations of at least some of the control sources on the corresponding ones of the indicators; and a circuit for detecting de-actuation of the pushbutton to select one of the control sources the representation of which is indicated on the corresponding one of the indicators at the time of the de-actuation.

Abstract: In a demodulator for a stereo signal transmitted by amplitude and phase modulation of a carrier, a phase signal and a quantity signal are formed. By multiplication of the quantity signal with the tangent of the phase signal reduced by a value to be preset, a difference signal is formed. The value to be preset is generated in such a way as to minimize the direct portion of the difference signal.

Abstract: A two channel audio switcher includes two analog switches, which may be implemented as a CMOS integrated circuit, receiving at their control input ports a complementary pair of drive signals, typically square wave, so as to toggle their outputs at a supersonic rate, which may be made variable. The switcher may receive as input two channels of a stereophonic source, two independent audio sources or a common monophonic source. The outputs typically drive an audio output unit such as the transmitter of a radio broadcasting system, an audio recording system, or amplifiers, controls and loudspeakers of a sound reinforcement system in an auditorium or studio. The system may be monophonic or stereophonic. In a stereophonic a.m. broadcasting system such as the Kahn independent sideband system, alternate switching allows each sideband to be modulated to full available power, thus allowing an increase in the apparent radiated power.

Abstract: In the case of a method for the generation of an amplitude-modulated ISB transmission signal from different AF signal (NF1,2), the signal vectors are split into signal components (x1,y1 and x2,y2, respectively), components sums (X,Y) are formed from the signal components and an amplitude signal (A(t)) and a harmonic phase signal are derived from the component sums. Amplitude signal and phase signal are then further processed separately and combined at the single transmitting tube (43) by combined amplitude modulation and phase modulation into the final transmission signal.

Abstract: Asymmetrical sideband amplitude modulation (AM) stereo transmission apparatus which transmits left (L) plus right (R) stereo information (L+R) in one stereo channel and a signal including L-R stereo information in a second stereo channel. Benefits include significant reduction of modulation envelope distortion in null regions of the transmitter's antenna radiation pattern and improved quality of reception in the presence of adjacent channel and co-channel interference, easing of the difficulty in tuning continuously tuned receivers and improvement of the fidelity of such receivers when side tuned, while at the same time providing a signal which, when received and decoded by an independent sideband AM stereo receiver, provides pleasing stereo effects.

Abstract: An improved asymmetrical sideband amplitude modulation (AM) transmitter is provided wherein the resulting transmitted signal produces a pleasing pseudo-stereo effect when received by a stereo receiver of the type designed to receive symmetrical sideband, phase-separated AM stereo radio braodcasts, when such receiver is operating in its stereo mode.

Abstract: This invention discloses an AM stereo transmission method characterized in that when left and right audio signals L and R are transmitted as an AM stereo signal which is represented by S=A cos(.omega.t+.phi.) wherein A denotes an amplitude, .omega. a carrier angular frequency and .phi. a phase angle given by tan.sup.-1 (L-R)/1+L+R, the audio signal of low frequency is transmitted at an amplitude A=(1+L+R), and the audio signal of high frequency is transmitted at an amplitude A=.sqroot.(1+L+R).sup.2 +(L-R).sup.2. The modulator according to the present invention comprises a matrix circuit for providing signals (L+R) and (L-R), an orthogonal modulator to produce an AM stereo signal .sqroot.(1+L+R).sup.2 +(L-R).sup.2 cos(.omega.t+.phi., an amplitude controller, an amplitude control signal generator circuit and a low-pass filter. In the modulator the said AM stereo signal S is changed to the signal (1+L+R)cos(.omega.t+.phi.) for audio signal of low frequency by controlling the amplitude controller.

Abstract: A system for transmitting a composite signal comprising a data transmission signal component and an AM broadcast signal component. The broadcast signal component may be monophonic or stereophonic. The level of the data signal component is made a function of the modulation level so that the data signal is masked by the program modulation and, therefore, AM radio listeners will not be disturbed by the data signal. The rate of data transmission, in one embodiment, is reduced as the level of the data signal is reduced. The data signal is in quadrature with the AM carrier so as to minimize detection of the data signal by an envelope demodulator. Suitable data receivers are also disclosed.

Abstract: The correction signal of the AM stereo system is filtered through a lowpass filter thus providing compatible signals at all normal modulation levels and frequencies, but allowing the signal to become pure quadrature at high frequency, high modulation levels. The inverse of the process can be utilized in the receivers if desired. The improvement is needed only for narrow channel or restricted sideband broadcasting.

Abstract: To permit transmission of traffic information by amplitude modulation in the medium wave band without disturbing audio signals being broadcast, a stereo pilot tone of frequency f.sub.pt below audible range is generated, as well as first and second sinusoidal signals of f.sub.1 =m.sub.1 /n.multidot.f.sub.pt and f.sub.2 =m.sub.2 /n.multidot.f.sub.pt, in which m.sub.1, m.sub.2, and n are different integers. The carrier is modulated by the stereo pilot tone and the first and second sinusoidal signals in such a manner that the phases thereof are different and further, when added, they do not essentially exceed the amplitude of the stereo pilot tone.

Abstract: In a system for processing stereo signals prior to application to an AM stereo channel information, the signals prior to broadcasting left channel and right channel automatic gain controllers are specially (L+R) gain controlled to maintain monaural loudness and are then followed by a low and high frequency equalizer and filter network. The system next transforms the audio into (L+R) and (L-R) signals and uses a multiband (L+R) and (L-R) peak limiting and clipping system with a variable gain (L-R) "stereo enhancement" amplifier located in the (L-R) path. The (L+R) and (L-R) audio is then applied to final peak clipping bandwidth filters. Finally, the (L+R) and (L-R) audio is transformed back into left and right audio which undergo left and right negative AGC'ing and peak clipping for decoding compatibility.