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: The invention provides an integrated circuit for reducing radio frequency interference due to antenna effects. In one embodiment, the integrated circuit includes a pre-stage audio processor and a post-stage power amplifier. The pre-stage audio processor receives a left channel input signal and a right channel input signal, processes the left channel input signal to obtain a positive left channel signal and a negative left channel signal, and processes the right channel input signal to obtain a positive right channel signal and a negative right channel signal. The post-stage power amplifier amplifies the positive left channel signal and the negative left channel signal to obtain a left channel output signal, and amplifies the positive right channel signal and the negative right channel signal to obtain a right channel output signal.

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: A method of denoise a stereo signal comprising a stereo sum signal and a stereo difference signal, performs a frequency selective stereo to mono blending based on the masking effect of the human auditory system. Therefore, a stereo signal noise reducer, comprising a first filter bank (1) to split the stereo difference signal (l?r) into a plurality of subbands, respective first multipliers (CO, . . . , CN) to weight each of the subbands of the stereo difference signal with a respective corresponding control signal (CO, . . . , CN), and a first adder (3) to sum all weighted subbands of the stereo difference signal (l?r) to build a frequency selective weighted stereo difference signal (diff), within which a number and width of the subbands obtained via the first filter bank (1) are choosen according to the properties of the human auditory system, further comprises a weighting factor determination unit which determines a respective control signal (CO, . . .

Abstract: A method and apparatus for processing an audio signal is disclosed. Herein, the method includes receiving a downmix information having at least two independent objects and a background object downmixed therein; separating the downmix information into a first independent object and a temporary background object using a first enhanced object information; and extracting a second independent object from the temporary background object using a second enhanced object information.

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 wireless communication device mixes audio content by combining audio signals from two or more different audio sources available at the wireless communication device and directing the mixed audio signal to a selected audio output circuit included in or associated with the wireless communication device. The audio signals can comprise received audio signals, stored audio signals, or a combination thereof. The audio signals can be synchronized, and one or more video signals may be synchronized with them. In at least one embodiment, the wireless communication device includes one or more circuits, such as a baseband or system processor, configured as a mixing circuit. An included (mixing) control circuit may be configured to control a user interface of the wireless communication device to support audio source selections, mixed audio signal output destinations, and audio signal control (volume, tone, etc.) for audio signals included in the mixed audio signal.

Abstract: A sound reproduction device such as a radio receiver or a cellular phone includes an antenna, a control unit and a speaker. The control unit determines whether a radio wave receiving condition is satisfactory. The receiving condition is represented by field strength of the radio wave or a degradation of a sound signal in the radio wave. If the receiving condition is not satisfactory, the control unit adds an acoustic effect to the sound signal thereby to notify listeners of degradation of receiving condition. The acoustic effect may be a changing of sound image localization, a changing of a frequency characteristic and a changing of a reverberation characteristic. The acoustic effect is stopped for a specified period after the acoustic effect is added, even when the receiving condition is not satisfactory.

Abstract: A method and apparatus is provided for remotely controlling a set of functions for a wireless audio system from a remote control. The method includes the steps of generating a digitally modulated pilot tone, and transmitting this pilot tone within a radio communication to a wireless receiver. The digitally modulated pilot tone carries data about a plurality of functions related to an audio source and its associated transmitter. The method further includes a step of receiving the data at a wireless receiver, augmenting it by information related to the wireless receiver, and communicating the resulting data via a communication network to a remote central control. The method further includes monitoring the resulting data at the remote central control, diagnosing detected problems associated with the wireless audio system, and communicating to the wireless audio system appropriate remedial action to alleviate the detected problems.

Abstract: A method of removing noise and interference from a signal by receiving the signal, calculating a joint time-frequency domain of the signal, estimating instantaneous frequencies of the joint time-frequency domain, modifying each estimated instantaneous frequency, if necessary, to correspond to a frequency of the joint time-frequency domain to which it most closely compares, redistributing the elements within the joint time-frequency domain according to the estimated instantaneous frequencies as modified, computing a magnitude for each element in the joint time-frequency domain as redistributed, plotting the results as the time-frequency representation of the signal, identifying in the plot any noise and interference components in the received signal, eliminating from the redistributed joint time-frequency domain elements that correspond to noise and interference, and recovering a signal devoid of noise and interference from the modified redistributed joint time-frequency domain.

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

Abstract: 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 system to generate an enhanced acoustic transmission signal includes a carrier signal generator to generate a carrier signal. A data signal generator is provided to receive data and to generate a data signal representing the data. A signal modulator is also provided to modulate the carrier signal with the data signal to form a modulated carrier signal at a carrier frequency. The system includes a masking signal generator to generate a masking signal to mask the modulated carrier signal from being audible by a human ear. An audio input device is provided to receive audio and to generate an audio signal based on the audio, wherein a frequency band surrounding the carrier frequency is removed from the audio signal. A signal adder is also provided to combine the modulated carrier signal, the masking signal, and the audio signal to form the enhanced acoustic transmission signal.

Abstract: An ad-hoc Adaptive Wireless Mobile Sound System, comprises a) a Master device, capable of sending a message/query to one or more mobile phone nodes, to verify whether they hold in their memory a specific song/track and to learn their music playing capabilities; b) transmission means provided in the Master, to convey to the mobile nodes that do not have a specific file in their memory, if any, a compressed audio stream of the said file; c) software suitable to convey, in a streaming fashion, the compressed audio file over wireless connectivity; d) software provided in the Master receives to select, based on an answer to said query, an appropriate surround scheme/algorithm configuration that takes into account the mobile nodes capabilities; e) communication means provided in the Master to publish the surround scheme/algorithm configuration in which a playing role/channel allocated to each of the mobile nodes; and f) means provided in the Master to synchronize the mobile nodes and to order them to play a stream

Abstract: A NICAM processing method includes receiving and temporarily storing a current frame of A-channel and B-channel input data into a first memory at a first clock rate. Companded A-channel and B-channel data of a previous frame are read from a second memory at a second clock rate and in a manner for interleaving the previous frame companded A-channel and B-channel data into the NICAM standard required interleaved format, wherein the companded A-channel and B-channel data of the previous frame was temporarily stored during a previous frame into the second memory in a format other than an interleaved format according to NICAM standard requirements.

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: In accordance with an example embodiment of the present invention, an apparatus is presented that comprises a multichannel audio transmitter which has a first mode for transmitting a multichannel audio signal and a second mode for transmitting at least a first and a second audio signal, each having fewer channels than the multichannel audio transmitter. A controller is configured to switch the multichannel audio transmitter from the first mode to the second mode when the first audio signal becomes available.

Abstract: Methods and units supporting a multichannel audio extension in a multichannel audio coding system are shown. In order to allow an efficient extension of an available mono audio signal {tilde over (M)} of a multichannel audio signal L/R, it is proposed that an encoding end of the multichannel audio coding system provides dedicated multichannel extension information for lower frequencies of the multichannel audio signal L/R, in addition to multichannel extension information at least for higher frequencies of the multichannel audio signal L/R. This dedicated multichannel extension information enables a decoding end of the multichannel audio coding system to reconstruct the lower frequencies of the multichannel audio signal L/R with a higher accuracy than the higher frequencies of the multichannel audio signal L/R.

Abstract: An apparatus for processing a media signal and method thereof are disclosed, by which the media signal can be converted to a surround signal by using spatial information of the media signal. The present invention provides a method of processing a signal, the method comprising of extracting spatial information and a downmix signal from a bitstream; and generating rendering information by using the spatial information and filter information having a surround effect, wherein the rendering information comprises first rendering information applied to one channel of the downmix signal extracted from the bitstream and then transmitted on the same channel and second rendering information applied to the channel and then transmitted on another channel.

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: There is disclosed a stereo encoding device capable of accurately encoding a stereo signal at a low bit rate and suppressing delay in audio communication. The device performs monaural encoding in its first layer (110). In a second layer (120), a filtering unit (103) generates an LPC (Linear Predictive Coding) coefficient and generates a left channel drive sound source signal. A time region evaluation unit (104) and a frequency region evaluation unit (105) perform signal evaluation and prediction in both of their regions. A residual encoding unit (106) encodes a residual signal. A bit distribution control unit (107) adaptively distributes bits to the time region evaluation unit (104), the frequency region evaluation unit (105), and the residual encoding unit (106) according to a condition of the audio signal.

Abstract: An encoding method and apparatus and a decoding method and apparatus are provided. The decoding method includes extracting a down-mix signal and down-mix identification information from an input bitstream, determining, based on the down-mix identification information, whether the down-mix signal is a 3D down-mix signal obtained by performing a three-dimensional (3D) rendering operation, and if the down-mix signal is not 3D down-mix signal, generating a 3D down-mix signal by performing a 3D rendering operation. Accordingly, it is possible to efficiently encode multi-channel signals with 3D effects and to adaptively restore and reproduce audio signals with optimum sound quality according to the characteristics of an audio reproduction environment.

Abstract: The present invention provides an audio processing circuit for receiving a first stream complying with a first standard (for example. MPEG audio standard) and generating a second stream complying with a second standard which is a digital interface standard (for example, S/PDIF standard). The audio processing circuit includes: a stream buffer for storing a plurality of frames of the first stream; a stream recovering circuit for detecting at least one of a plurality of fields in the frames, modifying at least one of the plurality of fields according to the first standard, and generating modified frames; a first buffer for storing the modified frames; and a burst circuit for partitioning the modified frames into a plurality of payload sections, adding a preamble to each of the payload sections, and forming the second stream.

Abstract: A method for processing a signal, wherein at least one of NF processing, audio processing, demodulation processing, and parts thereof of a processing section is suspended, while one or a plurality of certain suspending criteria are fulfilled, and wherein one or a plurality of suspending criteria are obtained or evaluated as control data from or by a pre-processing section with respect to the signal to be processed or a part or derivative thereof.

Abstract: The disclosure relates to communication devices which monitor an audio environment at a remote location and convey to a user a representation of that audio environment. The “representation” may be an abstraction of the audio environment at the remote location or may be a measure of decibels or some other quality or parameter of the audio environment. In some embodiments, the communication devices are two-way devices which allow users at remote locations to share an audio environment. In some embodiments, the communication devices are one way devices. In some embodiments, the communication devices may have the form of a window and be arranged to present sound in a manner that mimics sound received through a window. In such embodiments, the more open the window is, the more sound is relayed by the communication device.

Abstract: Method and system adapted to modifying an audio signal or speech signal comprising a step in which the frequency spectrum S(k) of the signal is converted by the application of a non-linear function. The method comprises at least the following steps: firstly, determining the signal level A(k), B(k) associated with a frequency k by taking account of different levels a(k), b(k) of the signal for the frequency k concerned and/or the neighboring frequencies (step 2a, step 8a); secondly, applying the non-linear function to said level A(k), B(k).

Abstract: The FM broadcast is received and the FM demodulated FM composite signal is stereo demodulated, and when the high band components of the demodulated signal are few, the correction is conducted by using the signals just before and just after the noise generation period, and when the high band components are large, the interpolation is conducted by using a central value calculated from the value of a predetermined period before the noise generation period, and a central value calculated from the value of a predetermined period after the noise generation period.

Abstract: Provided are an object-based three dimensional (3-D) audio service system using preset audio scenes and a method thereof. The system and the method are suggested for enabling a user to easily and conveniently watch and listen an object based 3-D audio service by eliminating inconvenience that requires a user to control each of object audio signals of sound sources. The system includes: audio input means for inputting an audio signal; preset audio scene generating means for extracting object audio signals from the audio signal inputted through the audio input means and generating more than one of 3-D audio scene information by arranging the extracted object audio signals in a 3-D space and editing features of each object; and encoding means for encoding and multiplexing the audio signal and the 3-D audio scene information for each object audio signal.

Abstract: The invention relates to the field of wireless communications and more specifically to an high quality, low power wireless audio system. More specifically, the invention comprises an audio source for receiving audio signals (e.g. music) and audio status information (e.g. song title) from a first external device (e.g. an MP3 player) and transmitting the audio signals and the audio status information over a wireless connection; and at least one audio sink for receiving the audio signals and said audio status information from the audio source and communicating the audio signals and the audio status information to a second external device (e.g. headphones), wherein a specified one of the at least one audio sink receives audio control information (e.g. pause) from the second external device and transmits said audio control information to said audio source via said wireless connection.

Abstract: In accordance with a method for providing a distinct perceptual location for an audio source within an audio mixture, a foreground signal may be processed to provide a foreground perceptual angle for the foreground signal. The foreground signal may also be processed to provide a desired attenuation level for the foreground signal. A background signal may be processed to provide a background perceptual angle for the background signal. The background signal may also be processed to provide a desired attenuation level for the background signal. The foreground signal and the background signal may be combined into an output audio source.

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 audio system for use in a passenger compartment of a vehicle. The system includes a front-driver side speaker unit and a front-passenger side speaker unit to cooperatively emit audio sound to the passenger compartment. The driver-side speaker unit includes a primary speaker and a secondary speaker to produce sound in cooperation with a primary speaker and a secondary speaker of the passenger-side speaker unit. An audio control system is configured to compensate for unbalancing of stereo images at front-driver/passenger positions by controlling output from the secondary speakers based on output from the primary speakers.

Abstract: A method for capturing information related to an audio signal includes broadcasting an acoustic signal capable of human perception, broadcasting an RF signal containing information related to content of the acoustic signal, receiving the acoustic signal, receiving the RF signal with a portable device, and using the portable device, storing the information related to the content of the acoustic signal. A similar method is used to capture information related to a visual display.

Abstract: A multi-channel signal decoding method is provided. A down-mixed signal representative of a multi-channel signal is decoded, and parameters representing characteristic relations between channels of the multi-channel signal are decoded. An additional parameter is estimated by using the decoded parameters, and the decoded down-mixed signal is up-mixed by using the decoded parameters and the estimated parameter so as to decode the multi-channel signal.

Abstract: A method and system for flexible FM tuning are provided and may include tuning to a particular frequency within a range of FM channels based on an IF frequency that includes an integer multiple of the channel spacing between neighboring allocated FM channels within the range of FM channels, offset by at most one-half the channel spacing. The method may further include determining whether the particular frequency comprises an on frequency channel, utilizing a frequency error that is based on the IF frequency. A local oscillator frequency may be selected for the tuning based on the frequency offset. An intermediate frequency (IF) channel may be generated utilizing the particular frequency and the selected local oscillator frequency. The generated IF channel may be between neighboring channels selected from the range of FM channels. The frequency error may be determined for the particular frequency within the range of FM channels.

Abstract: A demodulator (10) converts television signals to video baseband signals and audio baseband signals including stereo signals representing a right channel signal value, a left channel signal value and a pilot signal having an amplitude. A DSP (60) recursively finds a coefficient value for scaling that keeps the pilot signal amplitude within a range of values, there by improving baseband signal quality, such as stereo separation of stereo signals within the baseband signals.

Abstract: There is provided an audio signal transmitting apparatus including: a channel mapping unit which determines the audio signals allocated to each of the channels and generates channel mapping information representing types of the audio signals allocated to each of the channels; an audio signal transmitting unit which transmits the audio signals allocated to each of the channels by the channel mapping unit to the external connection apparatuses by a first transmitting scheme or a second transmitting scheme; and a connection management information providing unit which provides connection management information including the channel mapping information and transmitting scheme information representing the transmitting scheme of the audio signals to the external connection apparatuses.

Abstract: A stereo receiver of a simple structure is disclosed capable of continuously controlling the degree of separation of left and right voice signals in accordance with the level of electric field strength. The stereo receiver comprises electric field strength detecting means 9 for detecting the electric field strength of a received signal, and a stereo demodulation circuit 5 for outputting original voice signals of left and right channels. A blend circuit 6 is provided for blending the original voice signals with each other to output blended voice signals of the left and right channels. The degree of blending between the original voice signals is continuously changed correspondingly to the electric field strength. As a result, the S/N, which may deteriorate when the electric field strength is low, can be improved with a simple structure.

Abstract: A conference bridge of a conference system having a plurality of conference stations, the bridge comprising: reception means for receiving dual-channel audio signals coming from the conference station; determination means for determining at least one processing function per conference station of the conference system; application means for applying respective weighing functions to the received dual-channel signals; build-up means for building up one hybrid dual-channel signal for forwarding per conference station by means for summing a portion of the process dual-channel audio signals; and forwarding means for forwarding the respective hybrid dual-channel audio signal to each of the various conference stations of the conference system. A method of forwarding a spatialized audio scene and implemented by a conference bridge is also disclosed.

Abstract: A method for generating an encoded audio signal, and a method for processing the same during the multi-channel audio coding are disclosed. The present invention provides the method for generating an encoded audio signal comprising: generating basic spatial information including basic configuration information requisite for a multi-channel audio coding process and basic data corresponding to the basic configuration information; and generating extension spatial information including extension configuration information selectively required for the multi-channel audio coding process and extension data corresponding to the extension configuration information.

Abstract: A method for selecting an audio source when capturing audio data with at least one capture device includes connecting an audio source to an audio input port of the capture device, turning off the output of the audio source, calculating first signal strengths of the audio signals detected through the audio input ports of the capture device, turning on the output of the audio source, calculating second signal strengths of the audio signals detected through the audio input ports of the capture device, calculating a signal change between the first signal strength and the second signal strength of each audio input port, and selecting the audio input port with the largest signal change.

Abstract: Systems, methods and devices for gathering data concerning media usage by predetermined audience members. An audience measurement code in acoustic energy produced by a media receiver is detected by a portable media monitor. The monitor also detects a location code produced in a vicinity of the media receiver.

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: A method, apparatus and computer memory are provided for generating an audio fingerprint of an audio recording. A memory stores stable frequency family data corresponding to a plurality of stable frequency families. A processor curve fits audio recording data to at least one of the stable frequency families, extracts at least one variation from the curve fitted audio recording data, and creates the audio fingerprint of the audio recording from the at least one variation.

Abstract: A method of encoding a multi-channel audio signal including at least a first signal component (LF), a second signal component (LR) and a third signal component (RF). The method comprises the steps of encoding the first and second signal components by a first parametric encoder (202) resulting in a first encoded signal (L) and a first set of encoding parameters (P2); encoding the first encoded signal and a further signal (R) by a second parametric encoder (201), resulting in a second encoded signal (T) and a second set of encoding parameters (P1), where the further signal is derived from at least the third signal component; and representing the multi-channel audio signal at least by a resulting encoded signal (T) derived from at least the second encoded signal, by the first set of encoding parameters and by the second set of encoding parameters.

Abstract: Method for assigning a plurality of audio channels to a plurality of speakers, corresponding computer program product, storage means and manager node A method for assigning a plurality of audio channels of an audio data content to a plurality of speakers, each of the speakers being associated with a distinct node, called a speaker node, of a wireless network for the distribution of audio data contents, in the context of the diffusion of audio data content by a node, called a diffusion node, of the distribution network, the method being implemented by a node, called a manager node, of the distribution network.

Abstract: Apparatus and method for transmitting and reproducing stereophonic audio signals are disclosed. The method comprises splitting first and second channels into two frequency bands, combining the lower frequency band signals of the two channels, and transmitting the combined signals of the two channels or signals representative thereof. The apparatus comprises: a splitter for splitting each of the two channels into two frequency bands, a combiner, for combining the lower frequency band signals of the two channels, and a transmitter for transmitting the combined signals, or signals representative thereof, and the higher frequency band signals of the two channels, or signals representative thereof.

Abstract: The invention relates to an appliance (1) for converting DAB signals into FM signals and for transmitting the FM signals to an FM receiver (2), with a DAB receiver (3), which is equipped with an antenna input (4) and, at an output (10), emits an audio signal derived from a received DAB signal, and with an FM modulator unit (13) for generating an FM signal modulated with the audio signal, and a transmission unit for transmitting the FM signal to the FM receiver (2), and with a control stage (21), wherein the transmission unit is equipped with wired connection means (16), to which wired connection means (16) frequency detection means (20) are connected, which are connected to the control means (21) and arranged to detect the high frequency set in the FM receiver (2), and to generate a result signal representing the detection result and deliver it to the control stage (21) for channel setting purposes, and wherein the control stage (21) is connected to a channel-setting input (29) of the DAB receiver (3) in orde

Abstract: Apparatus and methods are disclosed for providing text information which identifies audio program selections, such as musical selections, stored on a compact disk (CD). Digital identification numbers encoded on the disk are provided by a compact disk player to a digital processor having a memory storage device. The identification numbers include an album identification number, and the track number selected for play. The memory storage device has stored therein text data identifying a plurality of program selections, organized by album and track identification number. The identification signals provided by the CD player are used to address the proper identifying text data for the selection being played. The identifying text data may be organized for radio broadcast along with the musical selection, and a receiver is used to display the data in text form to identify the musical selection as it is being reproduced.