Abstract: A digital audio encoder, digital video conditioner, and a digital modulator are described for producing a television broadcast signal at a desired channel frequency range. Left and right audio channel signals are digitized and encoded according to a stereo standard and then combined to form a stereo audio signal. A second audio programming channel signal may be included. A video input can be digitized and conditioned to form a digital video channel. The stereo audio signal can be placed directly at a desired channel frequency by frequency modulation without the need for using an intermediate frequency. The digital video channel can be placed at a desired frequency by amplitude modulation. The digital and audio channels can be digitally combined to create a television transmission signal at a desired frequency and according to a desired standard.

Abstract: A television audio signal encoder includes a matrix that sums a left channel audio signal and a right channel audio signal to produce a sum signal. The matrix also subtracts one of the left and right audio signals from the other to produce a difference signal. The encoder also includes a configurable infinite impulse response digital filter that selectively uses one or more sets of filter coefficients to filter the difference signal. Each selectable set of filter coefficients is associated with a unique filtering application to prepare the difference signal for transmission.

Abstract: A telephone line status notification system including at least one telephone line having a status, a communications network, at least one communications terminal which is connectable to the communications network and which is employable by a seeking user to communicate via the communications network a status request concerning the status of the at least one telephone line, apparatus for processing the status request the apparatus for processing is connectable to the communications network for receiving the status request from the seeking user therethrough and communicating the request, and apparatus for acquiring the status of the at least one telephone line, the apparatus for acquiring is in communication with the apparatus for processing for receiving the status request therefrom, and the apparatus for acquiring is connectable to the communications network for communicating the status via the communications network.

Abstract: A content processing device includes an input section, a memory section, a selection section and an output section. The input section is configured to receive content having a video signal and a plurality of audio signals with different audio types. The memory section is configured to store selection information indicative of a predetermined audio type, and order information indicative of a preset order of a plurality of audio types. The selection section is configured to select one of the audio signals based on the selection information and the order information. The output section is configured to output the video signal and the one of the audio signals that was selected by the selection section.

Abstract: An integrated digital BTSC encoder with an improved pilot signal generator substantially implemented on a single CMOS integrated circuit. By digitally generating a sinusoid that is frequency locked to a two-state input reference signal using a high rate internal clock, a hardware-efficient BTSC pilot signal generator is provided with good acquisition and tracking performance. Implemented efficiently as a simple phase detector, a low-complexity loop filter, a pilot frequency offset adder, a phase accumulator and a sinusoidal generator, the invention enables lower-rate post-processing of the pilot tone without a costly variable interpolator decimator structures.

Abstract: A sound signal processor and a method for processing a sound signal using the same enables the elimination of a plop noise generated in the course of a sound mode checking operation. The sound signal processor receives a sound intermediate frequency (SIF) signal and includes a channel mode discriminator for determining a current sound mode of the received SIF signal; a mode transition checker for performing a checking operation to determine whether the current sound mode has resulted from a transition from a first sound mode to a second sound mode; an amplitude measurement circuit for measuring an amplitude of the received SIF signal; and a controller for controlling a drive of the mode transition checker according to the measured amplitude. If the sound channel mode of a currently input sound signal is a mono channel mode, the amplitude of the input sound signal is measured, and the measured amplitude is compared to a reference level.

Abstract: A digital audio encoder, digital video conditioner, and a digital modulator are described for producing a television broadcast signal at a desired channel frequency range. Left and right audio channel signals are digitized and encoded according to a stereo standard and then combined to form a stereo audio signal. A second audio programming channel signal may be included. A video input can be digitized and conditioned to form a digital video channel. The stereo audio signal can be placed directly at a desired channel frequency by frequency modulation without the need for using an intermediate frequency. The digital video channel can be placed at a desired frequency by amplitude modulation. The digital and audio channels can be digitally combined to create a television transmission signal at a desired frequency and according to a desired standard.

Abstract: A television audio signal encoder includes an up-sampler that inserts additional samples into a sum signal, a difference signal, and/or a secondary audio program signal to increase the sample rate of the corresponding signal or signals. The sum signal represents the sum of a left channel digital audio signal and a right channel digital audio signal and the difference signal represents the difference between the left and right audio signals.

Abstract: System and method for synchronizing one or more secondary decoded media streams to a primary decoded media stream. The system includes a media stream processor and a mixer. The media stream processor receives a primary decoded media stream and secondary decoded media streams. The media stream processor synchronizes the secondary decoded media streams with the primary decoded media stream. The output of the media stream processor is coupled to the mixer. The mixer receives its second input from the primary decoded media stream. The mixer mixes the received streams and generates a PTS value for its output media stream by extrapolating the PTS of the primary decoded media stream.

Abstract: A digital audio encoder, digital video conditioner, and a digital modulator are described for producing a television broadcast signal at a desired channel frequency range. Left and right audio channel signals are digitized and encoded according to a stereo standard and then combined to form a stereo audio signal. A second audio programming channel signal may be included. A video input can be digitized and conditioned to form a digital video channel. The stereo audio signal can be placed directly at a desired channel frequency by frequency modulation without the need for using an intermediate frequency. The digital video channel can be placed at a desired frequency by amplitude modulation. The digital and audio channels can be digitally combined to create a television transmission signal at a desired frequency and according to a desired standard.

Abstract: A television audio signal encoder includes an up-sampler that inserts additional samples into a sum signal, a difference signal, and/or a secondary audio program signal to increase the sample rate of the corresponding signal or signals. The sum signal represents the sum of a left channel digital audio signal and a right channel digital audio signal and the difference signal represents the difference between the left and right audio signals.

Abstract: A television audio signal encoder includes a matrix that sums a left channel audio signal and a right channel audio signal to produce a sum signal. The matrix also subtracts one of the left and right audio signals from the other to produce a difference signal. The encoder also includes a configurable infinite impulse response digital filter that selectively uses one or more sets of filter coefficients to filter the difference signal. Each selectable set of filter coefficients is associated with a unique filtering application to prepare the difference signal for transmission.

Abstract: Aspects of the method and system for rejecting single sided leakage into an amplitude modulated channel may comprise an AM demodulator for demodulating a received composite broadcast signal's control channel to generate an amplitude modulated (AM) component. A PM demodulator may demodulate a received composite broadcast signal's control channel to generate a phase modulated (PM) component. At least one subtractor may subtract an AM leakage component and a PM leakage component contributed by the single sided leakage in the received composite broadcast signal's control channel. The system may comprise at least one of each of AM and PM bandpass filters and envelope detectors that filter and detect envelopes of the AM and PM components at 982.5 Hz and 922.5 Hz.

Abstract: A communication system including a transmitter, a receiver, and a TMDS-like link, in which video data and auxiliary data (typically including timing data associated with other auxiliary data) are transmitted from the transmitter to the receiver, or in which video data are transmitted over the link from the transmitter to the receiver, and auxiliary data (typically including timing data associated with other auxiliary data) are transmitted from the receiver to the transmitter. In typical embodiments the auxiliary data include one or more streams of audio data.

Abstract: An integrated digital BTSC encoder is implemented on a single CMOS integrated circuit chip with an all digital RF modulator. A set top box is provided that allows for a fully integrated solution which may be used with legacy television systems and also with other audio/visual equipment connected to the set top box.

Abstract: A television audio signal encoder includes a matrix that sums a left channel audio signal and a right channel audio signal to produce a sum signal. The matrix also subtracts one of the left and right audio signals from the other to produce a difference signal. The encoder also includes a configurable infinite impulse response digital filter that selectively uses one or more sets of filter coefficients to filter the difference signal. Each selectable set of filter coefficients is associated with a unique filtering application to prepare the difference signal for transmission.

Abstract: An integrated digital BTSC encoder with an improved pilot signal generator substantially implemented on a single CMOS integrated circuit. By digitally generating a sinusoid that is frequency locked to a two-state input reference signal using a high rate internal clock, a hardware-efficient BTSC pilot signal generator is provided with good acquisition and tracking performance. Implemented efficiently as a simple phase detector, a low-complexity loop filter, a pilot frequency offset adder, a phase accumulator and a sinusoidal generator, the invention enables lower-rate post-processing of the pilot tone without a costly variable interpolator decimator structures.

Abstract: System and method for an all-digital audio receiver for a BTSC MTS audio signal or other composite signal that is FM modulated. A preferred embodiment comprises a digital FM demodulator for receiving an analog to digital quantized SIF signal and performing demodulation and outputting a composite audio signal, and a digital audio processor for decomposing the composite audio signal into at least the SAP, stereo and monaural signals for audio reproduction. In a preferred embodiment, the digital audio processor is a programmable digital signal processor. In a preferred embodiment, the digital FM demodulator and the digital audio processor are implemented as an integrated circuit. Methods for processing the audio signal using the digital processors of the invention are provided.

Abstract: An integrated digital BTSC encoder with an improved pilot signal generator substantially implemented on a single CMOS integrated circuit is described. By digitally generating a sinusoid that is frequency locked to a two-state input reference signal using a high rate internal clock, a hardware-efficient BTSC pilot signal generator is provided with good acquisition and tracking performance. Implemented efficiently as a simple phase detector, a low-complexity loop filter, a pilot frequency offset adder, a phase accumulator and a sinusoidal generator, the invention enables lower-rate post-processing of the pilot tone without a costly variable interpolator decimator structures.

Abstract: A Japanese BTSC audio broadcast signal has three modes of transmission: mono, stereo, or dual mono. The control channel centering at 3.5 fH is one of the three channels in the J-BTSC signal, and contains information to indicate to the decoder which one of the three modes the audio transmission is in. The present invention uses a bandpass filter directly in the AM band, followed by envelope filtering and a decision circuit. Therefore, the need for AM demodulation and AM carrier detection is eliminated.

Abstract: An apparatus, method, and program for providing supplemental information related to a broadcast television program. Through a telephone or electronic network interface, a caller can request specific supplemental information related to one or more characters within the broadcast television program. The supplemental information is synchronized to the audio component of the television program so that there is no conflict, and is communicated to the caller via the network interface.

Abstract: A sampling rate converting circuit receives plural pieces of input data having different sampling frequencies. A plurality of FIR circuits is shared to reduce a circuit area, and, in a case where a magnification ratio of an input frequency and an output frequency is not an integer, signal deterioration due to resampling is solved. An oversampling component performs oversampling on input data Fs1 and outputs output data Fs?. An input timing timer calculates an input/input time based on an input timing signal CK1. An output timing timer and an accumulator calculate an input/output time based on an output timing signal CK? and the input timing signal CK1 and multiplies the input/output time by an oversampling multiple W to obtain a multiplied result. A divider divides the multiplied result by the input/input time to obtain a sampling position. A coefficient generator generates a filter coefficient based on the sampling position and supplies the filter coefficient to a multiplier.

Abstract: A BTSC encoder includes dual channel ADC, sync separator, audio processor, filtering device, and a composite audio signal generating device. The filtering device includes a first filter for providing a filtered L+R signal, and a second filter for providing at least one of: i) a filtered and combined pilot and modulated L?R signal and ii) separately filtered pilot and modulated L?R signals. The composite audio signal generating device is responsive to the filtered L+R signal, and at least one of i) the filtered and combined pilot and modulated L?R signal and ii) the separately filtered pilot and modulated L?R signals for generating and outputting a composite analog audio signal. In all embodiments, the modulated L?R signal is filtered via an anti-splatter filter.

Abstract: A method and apparatus are disclosed for performing sampling rate conversion of an audio signal from a first sampling rate to any of a plurality of higher sampling rates using a single set of low-pass filter coefficients. Sampling rate conversion is accomplished by effectively up-sampling, low-pass filtering, and down-sampling the audio signal to generate interpolated output samples of a second digital audio signal at any of a plurality of sampling rates. The sampling rate conversion process includes storing a fixed set of filter coefficients as a plurality of phased subsets of filter coefficients, applying samples of the audio signal to the phased subsets in a rotational manner to generate filtered samples of the audio signal, and selecting and linear interpolating between certain filtered samples to generate samples of the second digital audio signal.

Abstract: In a stereo/two-tone demodulator for demodulating an intermnediate-frequency signal having stereo or two-tone sound signals modulated on a first and a second sound carrier, the second sound carrier having a lower amplitude than the first sound carrier, ceramic filters or bandpass filters are saved and complete integration can be achieved in that a first (1) and a second (2) linear narrow-band phase-locked loop are provided each having their own frequency selectivity, the first narrow-band phase-locked loop (1) being implemented in such a way that it locks in at the frequency of the first sound carrier, and the second narrow-band phase-locked loop (2) being implemented in such a way that it locks in at the frequency of the second sound carrier.

Abstract: A remote control transmitter is provided with an AUDIO key for switching an audio mode in sound multiplex broadcasting and an ENTER key for determining various types of selections. When the ENTER key is operated in a state where the audio mode is temporarily selected by operating the AUDIO key, the state where the audio mode is temporarily selected is treated as the steady selection corresponding to setting on a menu screen.

Abstract: In a television receiving apparatus, a receiving apparatus 1 has storage means for storing a voice output switching table 20 registering a voice setting record 21 having the voice output format associated with the broadcasting language of the received digital broadcasting in a control unit 2. Every time the switching operation of the voice output format is made, the voice setting record 21 is selected in the registered order. Also, the receiving apparatus 1 outputs a voice signal with the language and the voice output format based on the selected voice setting record 21 from an output portion 6. Accordingly, the broadcasting language and the voice output format of the digital broadcasting being received can be switched by single operation, resulting in enhanced operability.

Abstract: Disclosed is a method and apparatus for transmission of audio signals over a dedicated video transmission line in which the analog audio signal is inserted into and becomes a part of the video signal. Samples of the audio signal are inserted into the active video line, either at the beginnings or endings, or both, of each active video line scan signal. A defined amount of the active video interval is reserved from the active video line, and is replaced with samples of the analog audio signal. An analog switch in the modulator of the invention toggles to pick up the samples of audio and place them at the designated location (either at the beginnings or at the ends) of each active video line scan signal. A corresponding switch in the demodulator of the invention toggles to retrieve the audio samples from the active video line scans for output to the user. Stereo and/or bi-directional audio signals can be transmitted by insertion into both the beginnings and the endings of the active video line scan signals.

Abstract: A video signal processor is used to receive programs in the form of a composite video signal which incorporates a video broadcast channel, a main audio channel and an additional audio channel. The main audio channel may carry a standard audio track for the program while the additional audio channel carries a different audio track, such as dialogue in a foreign language, DVS audio track, or a modified audio track that has been changed from the main audio track slightly. The video signal processor selects one of the audio tracks as the active audio track based on the contents of the additional audio channels and/or a viewers' preferences. The video signal processor may be a stand-alone device, or may be incorporated into a TV receiver, VCR, DVD player, etc.

Abstract: A circuit and method of interleaving a data stream (DATA) in which the data to be interleaved is contained in separate sections of the data stream. A buffer storage circuit (28) receives and stores the data stream. A first section (T1-T2) of the data stream is transferred to a first memory circuit (62) and a second section (T3-T5) of the data stream is transferred to a second memory circuit (64). A multiplexer circuit (68) receives data from the first and second memory circuits and selects between the first and second sections of the data stream in response to a selection signal to produce an interleaved output signal.

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: Disclosed is a method and apparatus for transmission of audio signals over a dedicated video transmission line in which the analog audio signal is inserted into and becomes a part of the video signal. Samples of the audio signal are inserted into the active video line, either at the beginnings or endings, or both, of each active video line scan signal. A defined amount of the active video interval is reserved from the active video line, and is replaced with samples of the analog audio signal. An analog switch in the modulator of the invention toggles to pick up the samples of audio and place them at the designated location (either at the beginnings or at the ends) of each active video line scan signal. A corresponding switch in the demodulator of the invention toggles to retrieve the audio samples from the active video line scans for output to the user. Stereo and/or bi-directional audio signals can be transmitted by insertion into both the beginnings and the endings of the active video line scan signals.

Abstract: Multiple signals received through different transmission mediums are synchronized within a set top box (STB) for subsequent mixing and presentation. For example, a video program transmitted via a cable network is synchronized with a secondary audio program, such as a radio program, received via the Internet. In another example, a video program transmitted via a cable network is synchronized with one or more programmed events, such as directives to display text or graphics or animation, play streaming media, direct a browser to a particular web page, and the like.

Abstract: According to one embodiment, in response to a user pressing an alternative audio button while watching a television broadcast, the user is provided with a list of secondary audio programs (e.g., radio programs received via the Internet) to replace the primary audio program associated with the broadcast. After a secondary audio program is selected, a synchronization component determines an extra transmission delay associated with the secondary audio program. A buffering component then buffers the video program for a period of time equal to the extra delay. A mixing component then mixes the synchronized video program with the secondary audio program, which are then presented to the user.

Abstract: A multilingual video-transmitting/receiving system includes multiplexing means for embedding a plurality of audio information channels in each of a plurality of video information channels and generating integral video signal, a plurality of subscriber receiving means for receiving the integral video signal, selecting one of video information channels from the integral video signal, and selecting one of the audio information channels embedded in the selected video information channel.

Abstract: In a method of calculating a variable spectral compression filter coefficient in a BTSC compatible stereo encoder from a feedback variable, a reciprocal value that is proportional to a reciprocal of the feedback variable is calculated. A lookup table is indexed using a preselected set of bits of the reciprocal value to generate at least one parameter of the filter coefficient function. An approximation of the compression filter coefficient is linearly interpolated based on the at least one parameter of the filter coefficient function. An apparatus that includes circuitry that calculates a variable spectral compression filter coefficient in a BTSC compatible stereo encoder from a feedback variable includes a reciprocal value circuit that calculates a proportional to a reciprocal value of the feedback variable.

Abstract: An integrated circuit for decoding an analog audio signal includes a tuner that receives the analog audio signal and provides a sound intercarrier frequency signal. A digital demodulator receives and digitizes the sound intercarrier frequency signal to provide a digitized sound intercarrier frequency signal, and digitally demodulates the digitized sound intercarrier frequency signal to provide a digitized multichannel television sound (MTS) demodulated signal. A digital broadcast television system committee (BTSC) compatible decoder receives and decodes the digitized multichannel television sound (MTS) demodulated signal, and provides a summed (L+R) audio output signal and a difference (L−R) audio output 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: 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: 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 includes a digital stereo modulator having a digital left channel input and a digital right channel input into a Digital BTSC Stereo Generator that contains a Sine Table. The Digital BTSC Stereo Generator generates and outputs the following signals: 2*FH, compressed BTSC L−R, preemphasized BTSC L+R, and FH. A digital multiplier amplitude modulates the 2*FH carrier with the compressed BTSC L−R output, and the resultant signal is a Stereo Subchannel AM-DSB-SC BTSC Compressed L−R, which is then summed together with the preemphasized BTSC L+R, and FH by a digital summer. The output of the digital summer is the composite BTSC output without SAP or Professional channels. The Digital BTSC Stereo Generator transforms left and right inputs into the outputs described above by utilizing clock inputs 3*FH and 12*FH. A digital PLL phase locks 3*FH and 12*FH clocks to a Horizontal Synch (FH) reference, for input to the Digital BTSC Stereo Generator.

Abstract: NTSC signals have a frame rate of 29.97 frames per second. In SDI, audio data is sampled at a rate 30n where n is an integer, e.g. 48K samples per second. Thus there are 1601.6 samples per frame. Although there are an integer number 8008 samples per five frame sequence, two frames have 1601 samples and 3 frames have 1602 samples. This creates a variable storage requirements from frame to frame and causes problems in editing. The SDI audio data is separated from associated video and decimated by a factor h, where h=4 for example in a signal processor 2, 3, 4, 5 and 6. The decimated data is written into a FIFO circuit at the rate 30n/h synchronously with frame timing under the control of a frame reference generator 10, 9. A fixed integer number n/h of samples is written into the FIFO 7 in each frame under the control of a write control circuit 8. n/h=400 for example, where n=1600 and h=4. Any additional sample in each frame is dropped. The 400 sample frames are stored on a disc 14.

Abstract: The present invention is a method and apparatus for providing stereo audio signals is disclosed. The circuit for providing stereo audio signals comprises an interface circuit coupled to receive an audio signal. The interface circuit has an analog-to-digital converter for digitizing the audio signal to provide a first audio input signal and a second audio input signal. A first processing circuit coupled to the interface circuit is configured to generate a first audio output signal based on the first audio input signal. A second processing circuit coupled to the interface circuit is configured to generate a second audio output signal based on the second audio input signal. The circuit further comprises a processor coupled to the interface circuit and a memory for storing instruction sequences.

Abstract: An apparatus for receiving broadcasting signals includes a tuner for receiving a digital audio broadcasting signal, a channel decoder for obtaining information data based on the digital audio broadcasting signal received by the tuner, a source decoder for causing the information data obtained from the channel decoder to be subjected to a decoding processing to produce a digital audio signal and associated data accompanying the digital audio signal, a random access memory in which the associated data are stored and from which the associated data stored therein are read, and a control unit operative to store a selected part of the associated data obtained from the source decoder, which has predetermined quantity of data, in the random access memory, to read the selected part of the associated data stored in the random access memory therefrom and to extract a specific portion of the associated data from the selected part of the associated data read from the random access memory.

Abstract: An apparatus for generating a broadcast television stereo signal has a left channel and a right channel. An analog-to-digital converter converts the right-channel signal to a right digital signal and the left-channel to a left digital signal. A circuit, which could include a digital signal processing chip, generates a first signal that includes the right digital signal added to the left digital signal and generates a second signal that includes the right digital signal subtracted from the left digital signal. The circuit also generates a digitally pre-emphasized first signal, which includes digital pre-emphasis, and a digitally pre-emphasized second signal, which also includes digital pre-emphasis. The circuit transforms the pre-emphasized first signal to a digital BTSC L+R signal and transforms the pre-emphasized second signal to a digital BTSC L-R signal.

Abstract: In a single-chip multistandard video modulator arrangement in which audio signals for transmission with video signals are first modulated onto a subcarrier and then combined with the video signals for modulation onto a VHF or UHF main carrier, the modulated subcarrier signals are passed through a high-pass filter before being combined with the video signals in order to reduce the amplitude of any audio frequency components, and the resulting variations in main carrier amplitude, which tend to produce "sound in vision" effects.

Abstract: In a bilingual television signal processor which is capable of automatically controlling a bandpass filter for detecting a pilot signal included in a bilingual signal received in a television receiver having a bilingual function without an external adjustment, an improvement including a microcomputer for outputting an on-screen display message indicating a bilingual broadcasting and scanning the locked information of the pilot signal outputted from the PLL detector in a predetermined voltage range and step and outputting a corresponding pulse width modulated (PWM) signal, and a digital/analog (DA) converter for converting the PWM signal outputted from the microcomputer into an analog signal and outputting a corresponding DC voltage to set the bandpass filter, can achieve the effect that it can be controlled without an external adjustment since the voltage for adjusting the center frequency of the bandpass filter is scanned in an appropriate range, and as a result, the shift in the center frequency of the band

Abstract: The present invention provides an apparatus that transmits the stereo and the surround sound information of the L-R signal without excessive attenuation at the higher frequencies in order to allow transmission of surround sound through a noisy medium. The apparatus includes a surround sound conditioner coupled to receive a L-R signal portion of the television signal for pre-emphasizing and companding the L-R signal to produce a conditioned L-R signal; a mixer coupled to the surround sound conditioner for switching the conditioned L-R signal at a predetermined value to produce an amplitude modulated L-R signal; and a synchronizing circuitry coupled to the mixer for maintaining the predetermined value in step with the reference value, wherein the mixed L-R signal is combined with a L+R signal portion of the television signal to produce a composite output signal.

Abstract: A method and system for broadcasting digital audio signals is disclosed. In one embodiment, the method includes the steps of processing at least a first audio signal from a first audio source, inserting at least the first audio signal onto the vertical blanking intervals of at least a first television broadcast signal, and transmitting at least the first television broadcast signal having at least the first audio signal inserted therein to customers.

Abstract: A component system includes a selector connected to at least one audio component, at least one video component includes a teletext receiver wherein audio and video signals output by the audio and video components as well as the teletext receiver can be selected, and when an operation to switch from an already selected component to the teletext receiver is carried out, a video signal produced by the teletext receiver is selected and output with an audio signal produced by the previously selected component remaining selected and output, allowing information transmitted by a teletext to be obtained without interrupting the audio signal being output by the previously selected component.