Abstract: A digital broadcasting system which is robust against an error when mobile service data is transmitted and a method of processing data are disclosed. The mobile service data is subjected to an additional coding process and the coded mobile service data is transmitted. Accordingly, it is possible to cope with a serious channel variation while applying robustness to the mobile service data.

Abstract: A method of processing audio information for broadcast to an audience can include changing first audio information from occurring in a first time interval to occurring in a second time interval to provide time-changed audio information. The time-changed audio information can be combined with second audio information that is responsive to the first audio information to provide broadcast audio information.

Abstract: A system for wireless transmission between a device such as a game controller and a base transceiver linked with a host device. Multiple controllers can be linked with the base transceiver through radio frequency transmission, and audio signals can be selectively transmitted therebetween. Audio commands can be introduced with manual controls at the controller for transmission to the base transceiver. Audio signals can be generated, mixed or otherwise processed by the combination of the host device and base transceiver for selective transmission to one or more controllers. The combination of manual switches, audio commands, and radio frequency transmission provides unique combinations of interaction between a person and the host device.

Abstract: Systems and techniques for processing signals include, in at least one aspect, a method including: receiving an input signal having an associated carrier frequency; multiplying the input signal to produce a multiplied signal; filtering the multiplied signal to produce a high frequency component of the multiplied signal; delaying the multiplied signal to produce a delayed signal; and combining the high frequency component of the multiplied signal with the delayed signal to produce a scaling factor for use in detecting at least one of multiple signal bitstreams associated with the input signal.

Abstract: The invention concerns a process for joint synthesis and spatialization of multiple sound sources in associated spatial positions, including: a) a step of assigning to each source at least one parameter (pi) representing an amplitude; b) a step of spatialization consisting in implementing an encoding into a plurality of channels, wherein each amplitude (pi) is duplicated to be multiplied to a specialization gain (gim), each spatialization gain being determined for one encoding channel (pgm) and for a source to be spatialized (Si); c) a step of grouping (R) the parameters multiplied by the gains (Pim), in respective channels (pg1, . . . , pgM), by applying a sum of said multiplied parameters (pim) on all the sources (Si) for each channel (pgm), and d) a step of parametric synthesis (SYNTH(I), . . . , SYNTH(M)) applied to each of the channels (pgm).

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: A transmission device includes a transmission power setting unit that respectively sets a transmission power for transmitting a main sound signal indicating a main sound and transmission powers for transmitting one or two or more subsidiary sound signals which pertain to the main sound signal and which realize a predetermined sound effect, and a first communication unit that respectively transmits the main sound signal and the one or two or more subsidiary sound signals on time division channels using the same frequency band, based on the transmission powers set by the transmission power setting unit, wherein the transmission power setting unit sets the transmission power for the main sound signal to a first reference value, and sets the transmission powers for the subsidiary sound signals to a second reference value smaller than the first reference value.

Abstract: A device for playing audio clips, a method of playing audio clips and a data storage medium having stored thereon computer code means for instructing a computer system to execute a method of playing audio clips. The device comprising a processor scalable in voltage, frequency, or both; a switch for selecting one of a plurality of output modes of the device; and a controller for controlling the processor to decode input joint/MS stereo mode encoded audio data representing said audio clip based on the selected output mode; wherein each output mode defines a number, m, of subbands of an M channel and a number, s, of subbands of an S channel of the joint/MS stereo mode encoded data; and the controller controls the processor to decode and store only data from the m and s subbands for playback.

Abstract: A method for automatically tuning a frequency modulator in a mobile device is described. A frequency band is automatically scanned using a frequency modulation (FM) receiver. The FM receiver is integrated as a part of the mobile device. Quality associated with channels of the frequency band is analyzed to identify at least one available channel at a first frequency. The first frequency is assigned to an FM modulator. The FM modulator is integrated as a part of the mobile device. A determination is made whether a command to scan for a second frequency is received. If the command to scan for the second frequency is not received, a signal on the first frequency is transmitted by the FM modulator.

Abstract: Separating a source in a stereo signal having a left channel and a right channel includes transforming the signal into a short-time transform domain; classifying portions of the signals having similar panning coefficients; segregating a selected one of the classified portions of the signals corresponding to the source; and reconstructing the source from the selected portions of the signals.

Abstract: An interface circuit for interconnecting the fixed controls of a vehicle to a replacement stereo. The interface circuit is capable of receiving control signals from fixed controls, such as steering wheel controls, backseat controls, handlebar controls, and the like, and then transmitting corresponding control signals to a replacement stereo. In one embodiment, the output signals from the interface circuit are comprised of wireless signals, such as IR signals. In another embodiment, the interface device is programmable such that the interface device can be programmed to emit signals corresponding to the same signals that would be emitted from a handheld remote control sold in conjunction with an after-market stereo receiver.

Abstract: A front-end processing circuit applies predetermined processing to an input signal and outputs a processed signal. A frequency modulation circuit applies frequency modulation to the output signal of the front-end processing circuit and outputs a processed signal. The front-end processing circuit includes a variable amplifier and a level detector. The variable amplifier receives a feedback signal and amplifies a signal by an amplification factor corresponding to the feedback signal. The level detector outputs the feedback signal to the variable amplifier in accordance with a level of a signal obtainable in a circuit succeeding the variable amplifier. The front-end processing circuit maintains the signal supplied to the frequency modulation circuit at a constant level.

Abstract: A noise level of an output signal is maintained low. An adjacent interference detecting unit compares a signal level of a wide band signal which is a desired station signal of a relatively wide band and a signal level of a narrow band signal which is a desired station signal of a relatively narrow band, and judges that adjacent interference has occurred when the signal level of the narrow band signal is less than or equal to a predetermined level even though the signal level of the wide band signal is greater than or equal to a predetermined level. A variable amplifier (36) attenuates an L?R signal and forms a monophonic signal when the signal level of the narrow band signal is lower than a predetermined level. With this process, when there is adjacent interference, a signal level of the narrow band signal at which the attenuating unit starts attenuating the L?R signal is changed to a relatively high value.

Abstract: The present invention provide a transmission system comprising a transmission apparatus for transmitting audio data of multi channels and auxiliary data required for playback of the audio data, and a receiving apparatus for receiving the audio data and the auxiliary data which are transmitted by the transmission apparatus. A multiplexer of the transmission apparatus creates block data that is composed of 8 frames, and first 1 byte of each frame is allocated to a header having Sync, OE and the like, the second byte is allocated to the auxiliary data including AUX data and copyright protect information, and remaining bytes are used to transmit the audio data. An encryptor carries out an encryption process for the second and later bytes of each frame, and a communication means outputs encrypted data.

Abstract: Electronic devices and equipment may communicate over a wired communications path. The wired communications path may include one or more wires and may be associated with a headphone cable. Data may be conveyed in the form of a digital data stream containing multiple traffic channels. The digital data stream may include superframes, each of which has multiple frames of data. The frames of data may each contain a number of data slots. Some of the slots in a superframe may be used exclusively by a particular one of the traffic channels. Boundary slots may be shared between traffic channels. Data interface circuitry may implement a data dispersion algorithm that determines the pattern in which data from each traffic channel is distributed within each boundary slot. Transmitting data interface circuitry may merge traffic channels into a single data stream. Receiving data interface circuitry may reconstruct the traffic channels.

Abstract: An apparatus for processing an encoded signal and method thereof are disclosed, by which an audio signal can be compressed and reconstructed in higher efficiency. An audio signal processing method includes the steps of identifying whether a type of an audio signal is a music using first type information, if the type of the audio signal is not the music signal, identifying whether the type of the audio signal is a speech signal or a mixed signal using second type information, and if the type of the audio signal is determined as either the speech signal or the mixed signal, reconstructing the audio signal according to a coding scheme applied per frame using coding identification information. If the type of the audio signal is the music signal, the first type information is received only. If the type of the audio signal is the speech signal or the mixed signal, both of the first type information and the second type information are received.

Abstract: Architecture for playing back personal text-based messages such as email and voicemail over a vehicle-based media system. The user can use a cell phone that registers over a cellular network to an IMS (IP multimedia subsystem) to obtain an associated IP address. The personal messages are then converted into audio signals using a remote text-to-voice (TTV) converter and transmitted to the phone based on the IP address. The phone then transmits the audio signals to the vehicle media system for playback using an unlicensed wireless technology (e.g., Bluetooth, Wi-Fi, etc.). Other alternative embodiments include transmitting converted message directly to the media system, via a satellite channel, converting the messages via a TTV converter on the cell phone, and streaming the converted messages to the phone and/or the media system for playback.

Abstract: The present invention relates to sound data transmission between a wireless sound transmitter that transmits sound data received from a multimedia source, and a wireless sound speaker that outputs sound signal by receiving the sound data, and more particularly, to a method for compensation of a replay time delay between the wireless sound speakers occurring when the multichannel sound data is distributed and transmitted from the wireless sound transmitter to the multiple wireless sound speakers. The differences in the replay time points occurring between the speakers in a TDMA based multichannel wireless transmission system, are pre-compensated through the delay of the sound data in the WSDT.

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 modulator apparatus comprising: an elongate housing having a distal portion and a proximal portion that flares outwardly from the distal portion; a cigarette lighter power adapter formed into the distal portion of the elongate housing; a user control panel located on the proximal portion of the elongate housing; an audio input jack in the elongate housing; a battery charging mechanism in the elongate housing for charging a battery of an external electronic device; a circuit located within the elongate housing and operably coupled to the cigarette lighter power adapter, the battery charging mechanism, the audio input jack, and the user control panel; and the circuit comprising means for transmitting an audio signal received by the audio input jack to a radio tuner.

Abstract: A lossless audio codec encodes/decodes a lossless variable bit rate (VBR) bitstream with random access point (RAP) capability to initiate lossless decoding at a specified segment within a frame and/or multiple prediction parameter set (MPPS) capability partitioned to mitigate transient effects. This is accomplished with an adaptive segmentation technique that fixes segment start points based on constraints imposed by the existence of a desired RAP and/or detected transient in the frame and selects a optimum segment duration in each frame to reduce encoded frame payload subject to an encoded segment payload constraint. In general, the boundary constraints specify that a desired RAP or detected transient must lie within a certain number of analysis blocks of a segment start point.

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: Methods and apparatuses for encoding and decoding a multi-channel audio signal are provided. In the encoding method, spatial information is calculated based on a multi-channel audio signal and a down-mix signal, and a compensation parameter that compensates for the down-mix signal is calculated based on the multi-channel audio signal and the down-mix signal. Thereafter, a bitstream is generated by encoding the spatial information, the compensation parameter, and the down-mix signal and combining the results of the encoding. Therefore, it is possible to prevent deterioration of the quality of sound regarding a multi-channel audio signal by compensating for the multi-channel audio signal using a compensation parameter that compensates for a down-mix signal.

Abstract: A connecting node in a first sub-network system reserves a transmission channel in a connecting network, reads waveform data having a designated signal name from a TL frame in the first sub-network system, and writes the read waveform data into the reserved transmission channel in a TL frame in the connecting network, and a connecting node in a second sub-network system reserves a transmission channel in the second sub-network system, reads the waveform data having the designated signal name from the TL frame in the connecting network, and writes the read waveform data into the reserved transmission channel in a TL frame in the second sub-network system.

Abstract: A parameter representation of a multi-channel signal having several original channels includes a parameter set, which, when used together with at least one down-mix channel allows a multi-channel reconstruction. An additional level parameter is calculated such that an energy of the at least one downmix channel weighted by the level parameter is equal to a sum of energies of the original channels. The additional level parameter is transmitted to a multi-channel reconstructor together with the parameter set or together with a down-mix channel. An apparatus for generating a multi-channel representation uses the level parameter to correct the energy of the at least one transmitted down-mix channel before entering the down-mix signal into an up-mixer or within the up-mixing process.

Abstract: A low-power, software based modulator that is used to transfer text data from an auxiliary audio device to an FM radio receiver includes a processor that receives the text data to be transmitted to the FM receiver, processes the data as an RDS signal or converts the text data to speech and then combines the processed data signal with an FM audio signal that is sent to the FM receiver. In one embodiment, the auxiliary audio device is a CD or CD-MP3 player and the software-based modulator is implemented on the control processor or DSP of the device. The FM encoder that produces the FM encoded audio signal may be a separate analog encoder or may also be implemented in the control processor of the auxiliary audio device.

Abstract: The present invention discloses a portable Karaoke system. The portable Karaoke system has a separate radio auxiliary microphone receiving module and a separate FM transmission module, which can be inserted/separated into/from the portable Karaoke system, such that a user can select and purchase only a required function module and easily carry and keep the portable Karaoke system. The portable system has hooks, which are provided at both sides of a battery case to prevent a battery from being easily separated from the battery case, and semicircular recesses, which are formed at the top edges of both sidewalls of the battery case to allow a user to easily remove the battery from the battery case. Furthermore, the portable Karaoke system has first and second gears formed at stepped portions of the end of a microphone body such that the first gear is combined with a general microphone and the second gear is combined with a ball-type microphone net.

Abstract: Provided are a method and apparatus for transmitting digital multimedia broadcasting data, and a method and apparatus for receiving digital multimedia broadcasting data. A basic audio signal and a multichannel audio signal are encoded to generate a basic audio stream and a multichannel audio stream, and a first data stream describing property and position data of the basic audio stream and a second data stream describing property and position data of the multichannel audio stream are transmitted as independent streams. According to the performance of the receiving apparatus, an audio signal may be decoded by using just the first data stream only or both the first data stream and the second stream.

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: The sound reproduction of a multichannel sound reproduction system with a plurality of speakers which is connected to the output of an FM stereo receiver is controlled by a control signal derived from the reception quality. Preferably, the control signal from the FM stereo receiver for controlling the stereo and mono components is also employed to control the multichannel sound reproduction system. For example, stereo, pseudo-stereo or mono reproduction are provided in the multichannel sound reproduction system in response to the stereo component within the output signal.

Abstract: A television audio signal encoder includes a device that sums a left channel audio signal and a right channel audio signal to produce a sum signal. The device 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. The set of filter coefficients is applied to the difference signal by a single multiplier in a recursive manner to prepare the difference signal for transmission.

Abstract: The invention relates to the analysis of characteristics of audio and/or video signalsforthe generation of audio-visual content signatures.To determine an audio signature a region of interest for example of high entropy —is identified in audio signature data. This region of interest is then provided as anaudio signaturewith offset information. A video signature is also provided.

Abstract: The invention discloses a decoding apparatus for decoding an analog audio signal. The decoding apparatus includes an RF tuner, an analog to digital (A/D) converter, a digital down converter, and a programmable digital signal processor (DSP). The RF tuner is used for receiving the analog audio signal and for providing an analog sound intercarrier frequency (SIF) signal indicative thereof. The analog to digital (A/D) converter is used for sampling the analog SIF signal and for converting the signal into a digital SIF signal. The digital down converter is used for down converting the digital SIF signal to generate a baseband signal. The programmable digital signal processor (DSP) is used for demodulating the baseband signal according to a demodulation procedure of a predetermined standard and decoding the demodulated baseband signal to output an output signal in compliance with a decoding procedure of the predetermined standard.

Abstract: A method and mobile device for delivering individual audio streams multiplexed in an audio data to at least one audio device through corresponding radio channels. The method includes establishing a communication channel with each of at least one audio device, receiving multiplexed audio data, extracting individual audio streams from the multiplexed audio data, allocating at least one of the individual audio streams to one or more communication channels, and transmitting the at least one of the individual audio streams to one or more of the at least one audio device through respective one or more communication channels.

Abstract: A stereo signal converter capable of realizing encoding with less redundancy, low bit-rate, and high quality even if the positions of sound sources are different from one another. In this device, a sample difference analyzing section (111) uses the signal in which a right-channel signal is shifted by a sample difference (d) in terms of time and a left-channel signal to compute a sample difference (D) in which the correlation becomes highest. A sample difference value computing section (112) computes a sample difference value (z) (the value to shift the right-channel signal in the current frame) on the basis of the value after the right-channel signal is shifted in the previous frame and the sample difference (D). A sample difference value encoding section (113) encodes the sample difference value (z). A slide section (114) shifts the right-channel signal by the sample difference value (z) in terms of time.

Abstract: A stereo audio decoder decoding a multiplex signal into a stereo audio signal. A first filter module filters the multiplex signal to generate a summation signal. A sub-carrier module modulates the multiplex signal according to a sub-carrier frequency to generate a sub-carrier mixed signal having a first high frequency and a first low frequency component. The first low frequency component has a sub-carrier phase offset between the stereo audio decoder and the multiplex signal. The second filter module filters out the first high frequency component of the sub-carrier mixed signal to generate a sub-carrier pure signal having only the first low frequency component. A corrector generates a difference signal according to a correction signal and the multiplex signal. A channel separator obtains a left channel signal and a right channel signal of the stereo audio signal by decoding the summation signal and the difference signal.

Abstract: In a system providing for delivery of audio signals from one or more source audio systems to two or more audio output sink systems, the sink systems establish duplex communication links (or bonds) to a bonding subsystem, which in turn is in duplex communication with the source audio systems. Data signals may be sent from sink systems, via the bonding subsystem, to request delivery of audio signals from selected audio source channels. In preferred embodiments, the sink systems may also request specific treatments of or modifications to audio signals to be sent to individual sinks in the sink systems. The bonding subsystem receives such treatment requests from the sink system and relays them to a matrix manager which initiates and controls the processing of source audio channels as required to comply with requests from the sink system. The matrix manager then delivers the treated or modified audio output channels to the bonding subsystem for delivery to the requesting sink systems.

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: Systems and techniques for digital processing of FM signals include, in at least one aspect, an FM digital processing method including receiving one or more digital signals including a first signal having a first frequency; obtaining a second signal by multiplying the first signal by X to assist in information recovery from the one or more digital signals based on the first signal; filtering the second signal to obtain a high frequency component of the second signal; delaying the second signal to obtain a delayed signal; and generating an output signal based on the high frequency component of the second signal and a normalization factor derived at least in part from the delayed signal.

Abstract: A mobile communication terminal is provided comprising a receiver for receiving broadcast content over a first broadcast channel; and a controller for controlling reproduction of the broadcast content on a display of the mobile terminal. In response to detecting a first event, the controller receives a first request to delay reproduction of broadcast content in real-time. In response to receiving the first request the controller stores, in a storage medium, first broadcast content received between a time T1 and T2. In response to detecting a second event, the controller reproduces the first broadcast content. The controller stores second broadcast content received after a time T3. After the first broadcast content is reproduced, the controller reproduces the second broadcast content.

Abstract: A stereo decoding system comprises an oscillator, a bandpass filter, a PLL unit and a stereo decoder. The oscillator generates a first signal with a center frequency. The bandpass filter receives a stereo multiplexed signal and the first signal to filter out a pilot signal. The PLL unit receives the pilot signal to generate a PLL output signal. The stereo decoder receives the stereo multiplexed signal and the PLL output signal to separate a left channel signal and a right channel signal from the stereo multiplexed signal.

Abstract: A television audio signal encoder includes a device that sums a left channel audio signal and a right channel audio signal to produce a sum signal. The device 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. The set of filter coefficients is applied to the difference signal by a single multiplier in a recursive manner to prepare the difference signal for transmission.

Abstract: A process for the stereophonic acquisition, transmission, and reproduction of sound events for communication applications in telephony with headphones and microphones available for each participant which to each ear area of every participant is assigned a combination consisting of an earphone or a headphone as well as a microphone each, provided in close proximity one to the other in connection that has no acoustic echo or feedback, whereby the real, binaurally acquired environment, related to the head of the respective acquiring participant with respect to its reflection, diffraction, and resonance behavior is transmitted to each of the participants in the form of corresponding stereophonic sound and acoustic images via a two-channel connection.

Abstract: In a mixing system where a console, engine and I/O devices, each possessing a plurality of supply sources and destinations, are connected to a network, a plurality of transmission channels of the network are allocated to the individual devices. In response to human operator's connection instruction given via the console for instructing a connection between a supply source possessed by one of the devices and a supply destination possessed by another of the devices, the device possessing the supply source transmits an audio signal of the supply source to the network using one of the transmission channels allocated to the device. The device possessing the signal supply destination, for which the connection instruction has been given, receives the audio signal transmitted via the transmission channel and supplies the received audio signal to the supply destination.

Abstract: In one embodiment, a method and apparatus of multiplexing and demultiplexing radio channels includes receiving data through at least one multicast media channel available for use in an Internet Protocol Interoperability and Communications System (IPICS) comprising multiple communication endpoints linkable to a Land Mobile Radio (LMR) in the IPICS; channel mapping the at least one multicast media channel to multiple media channels of the LMR; receiving an audio signal through the at least one multicast media channel; controlling access to the LMR by applying communication access control policies based on the received data upon reception of the audio signal; and operatively linking the LMR to a specified endpoint through the at least one multicast media channel based on the communication access control policies.

Abstract: A microphone system including a main unit for controlling the entire system and microphones having cascade connections from the main unit assuming the main-unit side upstream and the opposite side downstream. The microphone includes communication control means for controlling data transmitted between the main unit and microphones, sound input means for converting collected sound into a digital signal, echo cancellation means for eliminating an echo component in the sound signal, and sound-information generation means for updating the sound information by adding the sound signal of the self-microphone to the sound information of the downstream microphones and upstream transmitting the up data including the updated sound information. The microphone transmits the down data transmitted from the main unit to the down-most microphone in sequence in accordance with the cascade connection and transmits the up data from the down-most microphone to the main unit in reverse sequence.

Abstract: A stereo decoder includes a first digital filter for producing a first filtered composite audio signal and a second digital filter for producing a second filtered composite audio signal. A sum/difference network produces a left channel signal and a right channel signal. A processor executes operational instructions that calculate a first corner frequency of a first digital filter based on a signal quality of the FM signal, calculate a first set of filter coefficients for the first digital filter based on the first corner frequency and an approximation of a filter transform, calculate a second corner frequency of a second digital filter based on the signal quality, and calculate a second set of filter coefficients for the second digital filter, based on the second corner frequency and the approximation of the filter transform.

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: Estimates of spectral magnitude and phase are obtained by an estimation process using spectral information from analysis filter banks such as the Modified Discrete Cosine Transform. The estimation process may be implemented by convolution-like operations with impulse responses. Portions of the impulse responses may be selected for use in the convolution-like operations to trade off between computational complexity and estimation accuracy. Mathematical derivations of analytical expressions for filter structures and impulse responses are disclosed.