Abstract: A headphone socket that includes: a headphone socket body and a spring plate part; the spring plate part is clamped into the headphone socket body, and the spring plate part includes: a microphone spring plate, a grounded spring plate, a right-earpiece spring plate, a charging detection availability identifying spring plate, a headphone availability detection spring plate, and a left-earpiece spring plate; and when a headphone plug of headphones is completely inserted into the headphone socket, the charging detection availability identifying spring plate is in contact with a right-channel terminal of the headphone plug, so as to conduct a connection between the right-channel terminal and an external circuit, so that a device at which the headphone socket is located charges the headphones by using the microphone spring plate that is in contact with a microphone terminal of the headphone plug.

Abstract: A headset, which may be configured to process audio for a player participating in an online multiplayer game, may support auto-pairing with audio basestations. In this regard, an audio basestation may be utilized in communicating game and chat audio content from an associated game console to a paired headset and in communicating microphone audio from the paired headset to the console. The auto-pairing may comprise automatically exchanging signals with an audio basestation, in response to the headset coming within a determined distance of the audio basestation; and pairing with the audio basestation based on processing of the exchanged signals. The pairing may result in establishing one or more wireless links for game and chat audio to be communicated to the headset and microphone audio to be communicated from the headset. The determined distance may correspond to physical contact (e.g., tapping) between the headset and the basestation.

Abstract: There is provided an audio encoder comprising a determination part determining, based on frequency spectra of audio signals of a plurality of channels, a mixing ratio as a ratio, relative to a frequency spectrum after mixing for each channel of the plurality of channels, of the frequency spectrum for another channel, a mixing part mixing the frequency spectra of the plurality of channels for each channel based on the mixing ratio determined by the determination part, and an encoding part encoding the frequency spectra of the plurality of channels after mixing by the mixing part.

Abstract: Technology is described for providing audio for digital content. The digital content may be provided to a plurality of devices. Each device may be associated with a profile. At least one group of profiles may be identified from the plurality of devices that share an affiliation. The group of profiles may be represented as a group of environment objects in the digital content. A location may be identified within the digital content that corresponds to the group of environment objects. Audio may be received for the devices. The audio may be received while the digital content is being transmitted to the plurality of devices. The audio may be transmitted to the plurality of devices for directional audio playback. The audio may be directed to correspond with the virtual location of the at least one group of environment objects in the digital content.

Abstract: Suppression of interference across transceivers integrated on a single semiconductor chip. An example of a method of reducing noise in a transceiver includes introducing an adjustable time delay into a signal between a first section of a signal path into which noise may be introduced and a second section of the signal path into which noise may be introduced. The method also includes selectively adjusting the time delay and signal polarity to improve a signal-to-noise metric of the transceiver. An example of the transceiver includes a transmitter and a receiver. The transceiver also includes an adjustable time delay between a first section of a transceiver signal path into which noise may be introduced and a second section of the transceiver signal path into which noise may be introduced and circuitry for reducing noise by adjusting a value of the time delay.

Abstract: The present invention relates to a method and device for encoding or decoding an object audio signal or rendering the object audio signal in a three-dimensional space. The method for processing an audio signal, according to one aspect of the present invention, comprises the steps of: generating a first object signal group and a second object signal group obtained by classifying a plurality of object signals according to a determined method; generating a first down-mix signal for the first object signal group; generating a second down-mix signal for the second object signal group; generating first object extraction information in correspondence with the first down-mix signal with respect to object signals included in the first object signal group; and generating second object extraction information in correspondence with the second down-mix signal with respect to object signals included in the second object signal group.

Abstract: An apparatus for processing an audio signal and method thereof, the method including receiving a downmix signal including at least one normal object signal, and a bitstream including object information determined when the downmix signal is generated; extracting, from an extension part of the bitstream, an extension type identifier indicating whether the downmix signal further includes a multi-channel object signal; generating spatial information using the object information and mix information when mode information indicates that the multi-channel object signal is to be suppressed and the extension type identifier indicates that the downmix signal further includes the multi-channel object signal; and transmitting spatial information. The mix information is to control an object position or an object level of the at least one normal object signal.

Abstract: A stereophonic signal is converted into a mid channel signal and a side channel signal. Noise is added to the side channel signal. The amount of noise is selected depending on masking thresholds for at least two channels of the stereophonic signal. The mid channel signal and the modified side channel signal are quantized for transmission. Alternatively or in addition, a set of quantization parameter for the quantization of the side channel signal is selected depending on the masking thresholds.

Abstract: An apparatus for processing an audio signal and method thereof, the method including receiving a downmix signal including at least one normal object signal, and a bitstream including object information determined when the downmix signal is generated; extracting an extension type identifier indicating whether the downmix signal further includes a multi-channel object signal, from an extension part of the bitstream; extracting spatial information from the bitstream when the extension type identifier indicates that the downmix signal further includes the multi-channel object signal; and transmitting spatial information when mode information indicates that the least one normal object signal is to be suppressed. The spatial information is determined when a multi-channel source signal is downmixed into the multi-channel object signal. The mix information is to control an object position or an object level of the at least one normal object signal.

Abstract: A method for pre-processing a channelized music signal to improve perception and appreciation for a hearing prosthesis recipient. In one example, the channelized music signal is a stereo input signal. A device, such as a handheld device, hearing prosthesis, or audio cable, for example, applies a mask to a stereo input signal to extract a center-mixed component from the stereo signal and outputs an output signal comprised of a weighted combination of the extracted center-mixed component and a residual signal comprising a non-extracted part of the stereo input signal. The center-mixed component may contain components, such as leading vocals and/or drums, preferred by hearing prosthesis recipients relative to other components, such as backing vocals or other instruments.

Abstract: An encoder/decoder is based on a combination of two audio or video channels to obtain a first combination signal as a mid-signal and a residual signal derivable using a predicted side signal derived from the mid-signal. A decoder uses the prediction residual signal, the first combination signal, a prediction direction indicator and prediction information to derive decoded first channel and second channel signals. A real-to-imaginary transform can be applied for estimating the imaginary part of the spectrum of the first combination signal. The prediction signal used in the derivation of the prediction residual signal, the real-valued first combination signal is multiplied by a real portion of the complex prediction information and the estimated imaginary part of the first combination signal is multiplied by an imaginary portion of the complex prediction information.

Abstract: Reduced digital audio sampling rates are described in a digital audio processing chain. In one embodiment, an audio signal is received. A convolution operation is performed on the received audio signal. The convoluted audio signal is sampled. An interrupt is received to process the audio sample, and the sample is processed in response to the interrupt. The processed samples are collected to form a frame and the frame is transmitted to a remote device.

Abstract: Systems and methods are disclosed that may be implemented to process a received RF spectrum that includes both analog modulated and digitally modulated RF signals to blend between a digital demodulated signal and an analog demodulated signal obtained from the received RF spectrum prior to performing one or more signal quality mitigation operations on the blended signal (e.g., such as stereo blend, hi-cut, etc.). In one embodiment, the digital demodulated signal and the analog demodulated signal may include at least some of the same information, e.g., such as information from simulcast digital and analog channels that are obtained from the same received RF spectrum.

Abstract: The invention relates to a method and an apparatus in which samples of at least a part of an audio signal of a first channel and a part of an audio signal of a second channel are used to produce a sparse representation of the audio signals to increase the encoding efficiency. In an example embodiment one or more audio signals are input and relevant auditory cues are determined in a time-frequency plane. The relevant auditory cues are combined to form an auditory neurons map. Said one or more audio signals are transformed into a transform domain and the auditory neurons map is used to form a sparse representation of said one or more audio signal.

Abstract: The broadcasting of audio contents by two loudspeakers is controlled by delivering a first audio content to the two loudspeakers and a further processing in which an auxiliary audio content is received. A second audio content is formed by temporally delaying the auxiliary audio content with a delay dependent on the spacing between the loudspeakers and on a distance between a first loudspeaker and a spot located in front of this first loudspeaker. The second audio content is delivered to the first loudspeaker. A third audio content is formed by inverting the auxiliary audio content. The third audio content is then delivered to the second loudspeaker.

Abstract: An apparatus comprising at least one processor and at least one memory including computer program code the at least one memory and the computer program code configured to, with the at least one processor, cause the apparatus at least to perform: determining from an audio signal at least a first part and a second part; encoding the first part of the audio signal with a first encoder for generating a first encoded audio signal; encoding the second part of the audio signal with a second encoder configured to generate a second encoded audio signal comprising for a first section of the second part an indicator to at least part of the first part of the audio signal; and determining the first section of the second part of the audio signal such that the first encoded audio signal and second encoded audio signal is within a defined encoding efficiency parameter.

Abstract: A system for adaptive bit rate distribution of multicast streams may include one or more processors and a memory. The one or more processors may be configured to identify streams, such as multicast streams, transmitted by a content delivery network. The streams may each contain a content item that is encoded at a different bit rate for each stream. The one or more processors may determine segments of the streams, for example based on time stamps associated with the content item or markers of the streams. The one or more processors may transmit a first segment of a first stream encoded at a first bit rate to a user device in response to a request therefor and, upon completing the transmission of the first segment, transmit a second segment of a second stream encoded at a second bit rate to the user device in response to a request therefor.

Abstract: The present invention aims to provide a resource allocation method in a communication system that can effectively suppress a transmission delay. Provided is a resource allocation system in a communication system which includes a terminal 110 and a base station 100, wherein data is regularly generated at the terminal, the base station allocates a resource to the terminal, and the terminal transmits data by using the resource. The terminal 110 includes a generated traffic monitoring unit 113 that monitors the data and, when a predetermined change occurs in the data, reports it to the base station. The base station 100 includes a resource management unit 102 that determines a resource allocation according to the report from the terminal.

Abstract: Apparatus to reduce noise in a stereo FM broadcast received via an antenna, the apparatus comprising: a frequency translator configured to translate the received stereo FM broadcast to an intermediate carrier frequency, a demodulation unit configured to demodulate the translated FM signal so as to form left-plus-right and left-minus-right AM signals, a filter configured to form a filtered signal by filtering one of the AM signals, so as to suppress a sub-band of that signal containing unwanted tones, a summing unit for summing the filtered signal and the other of the left-plus-right and left-minus-right signals to produce a stereo audio signal.

Abstract: A method, computer-readable storage medium, and signal processing apparatus for processing a plurality of input signals. The method includes receiving or generating a first intermediate signal and a second intermediate signal. The first and second intermediate signals are summed and the summed signals are output to a signal analog-to-digital converter having a predetermined sampling frequency.

Abstract: Receivers accommodating carrier frequency selection methods in wireless communications systems employing multiple carrier frequencies are described. Although the receiver is tuned to a single band, an estimate of the channel quality corresponding to the currently used carrier and an alternative carrier is generated without switching between carriers. Transmitters of different cells and/or different sectors primarily use different carrier frequencies but periodically transmit using a neighboring sector's carrier frequency. Mobile node receivers use a single RF chain with a controllable RF filter to receive and process a signal within a first selected carrier band including two components, a first signal component identified with the first currently selected band and a second signal component identified with a second alternative band.

Abstract: Systems, apparatuses and methods for providing FM phase diversity capabilities are provide. In some aspects, a vehicle system includes a first FM antenna positioned externally of a vehicle and adapted to receive a first FM signal, a radio unit positioned internally of the vehicle and including a display and at least one user control, a cable electrically coupled to the first FM antenna and the radio unit to communicate the first FM signal to the radio unit, and a second FM antenna positioned internally of the vehicle and adapted to receive a second FM signal that is communicated to the radio unit. In another aspect, a radio unit positioned internally of the vehicle may include a printed wire board and a second FM antenna may be positioned on the printed wire board internally of the vehicle and adapted to receive a second FM signal.

Abstract: Systems and methods are provided for adding audio content to a playback queue. In one example, a method involves receiving a request to add automated streaming audio content to a particular playback position in a playback queue. The playback queue may indicate a plurality of audio content and a corresponding playback position associated with each of the plurality of audio content for playback. The method further involves determining a playback duration for the automated streaming audio content, adding the automated streaming audio content to the playback queue at the particular playback position, and causing the automated streaming audio content to be rendered at the particular playback position in the playback queue for the determined playback duration. The automated streaming audio content may include discrete automated streaming audio content and/or continuous automated streaming audio content. Respective examples for adding discrete and continuous automated streaming audio content are also provided.

Abstract: The automated switching of broadcast sources of an audio content stream based on the geographic location of the physical unit that is receiving the audio content stream. For instance, while rendering a particular audio content stream from a first broadcasting source, a system may have an identity of one or more alternative possible broadcasting sources available for the same audio content stream for that relative geographic location. In response to some decision making, the system may decide to switch broadest sources for that audio content stream. Thus, the user continues to listen to the audio content stream without being subject to possible negative consequences of continuing to listen to the audio content stream as received from the original broadcast station. Data transmitted by the broadcast stations themselves need not be relied upon. Thus, the automated switching is accomplished even outside of regions in which broadcast stations transmit such data.

Abstract: An apparatus for enhancing a multichannel audio signal comprising at least two channels configured to: estimate a value representing a direction of arrival associated with a first audio signal from at least a first channel and a second audio signal from at least a second channel of at least two channels of a multichannel audio signal; determine a scaling factor dependent on the direction of arrival associated with the first audio signal and the second audio signal; and apply the scaling factor to a parameter associated with a difference in audio signal levels between the first audio signal and the second audio sign.

Abstract: A digital audio data output device having first and second post-processors that process received digital audio data in parallel. Each processor includes a mixer unit that processes received digital audio data on the fly in response to a channel mode control signal, and a volume control unit that adjusts volume level of output of the mixer unit on the fly in response to a volume level control signal. Each mixer unit includes first and second buffers that can respectively store first and second channel data, a calculator that calculates mix data from the outputs of the buffers, a third buffer that stores the mix data, and an output unit. The output unit selects and outputs one of the received digital audio data, the output of the first buffer, the output of the second buffer, the mix data, and output of the third buffer, as the output of the mixer unit.

Abstract: The invention relates to a method for improving a stereo audio signal of an FM stereo radio receiver. The method comprises determining one or more parametric stereo parameters based on the stereo audio signal in a frequency-variant or frequency-invariant manner. Preferably, these PS parameters are time- and frequency-variant. Moreover, the method comprises generating the improved stereo signal based on a first audio signal and the one or more parametric stereo parameters. The first audio signal is obtained from the stereo audio signal, e.g. by a downmix operation.

Abstract: A method and apparatus is disclosed in which delay is applied to analog-sourced audio in a radio simulcast when the analog signal initially leads the digital signal. A radio receiver is configured to receive a simulcast radio program broadcast with an analog signal and a digital signal. The program content can be extracted from either the analog or digital signals, with the audio source eventually being blended to the digital signal. Audio is initially provided based on the analog signal. If the analog signal is initially leading the digital signal, delay is applied a data stream corresponding to the analog signal relative to a data stream corresponding to the digital signal. Delay applied to the data stream corresponding to the analog signal is increased at a rate that avoids audio artifacts of the output audio. The blend is performed when the data streams are aligned in time.

Abstract: A method and apparatus are provided for generating a personalized radio channel playlist by simultaneously buffering tracks from multiple received channels from one or more source streams and selectively playing back tracks from the buffered channels. Navigation tools permit users to skip buffered songs in their playlist (e.g., skip forward and backward). Users can specify favorite channels for building personal playlists, or multiple default playlist channels can be provided (e.g., by genre). Thumbs up/down buttons on the radio receiver permit entering a song or artist being played back into a favorites list that is used to search all channels for matches or a banned list used to block songs from future playlists. A matched channel carrying the favorite can be added to a playlist. Segments on the playlist can be played back in full or truncated to facilitate preview of playlist segments.

Abstract: A wireless speaker device includes a speaker module, a processing module, a wireless module and a control module. The processing module may receive the mate signal sent by the adjacent wireless speaker device through the wireless module or send the mate signal sent to the adjacent wireless speaker device through the wireless module to establish a wireless multi-channel audio system. Therefore, the connections of the wireless speaker devices within the wireless multi-channel audio system are simplified. Ans the cost time to mate for connection is reduced. Also, the connection is automated to be convenient for users to operate.

Abstract: A system for wirelessly transmitting and receiving customized data broadcasts comprising at least one computer apparatus running an application for customized data broadcasts. The computer apparatus uses the application to transmit and/or receive at least one broadcast. The system is further comprised of a server network infrastructure with: the application, at least one application server, at least one web server; and at least one database/information server. The infrastructure, servers, and application comprise a distributed computing environment in a secure server network model wherein the environment utilizes the application as a secure communicator of the broadcasts, thereby enabling secure communication of the broadcasts between any computer apparatus running the application.

Abstract: Disclosed is an audio encoding device which removes unnecessary inter-channel parameters from the subject to be encoded, improving the encoding efficiency thereby. In this audio encoding device, a principal component analysis unit (301) converts an inputted left signal {Lsb(f)} and an inputted right signal {Rsb(f)} into a principal component signal {PCsb(f)} and an ambient signal {Asb(f)} and calculates for each sub-band, a rotation angle which indicates the degree of conversion; a monophonic encoding unit (303) encodes the principal component signal {Pcsb(f)}; a rotation angle encoding unit (302) encodes the angle of rotation {?b}; a local monophonic decoding unit (603) creates a decoded principal component signal; and a redundant parameter elimination unit (604) identifies the redundant parameters by analyzing the encoding quality of the decoded principal component signal and eliminates the redundant parameters from the signal to be encoded.

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: A method of optimizing stereo reception for an analog radio by applying the demodulated right sound signal (SD) and left sound signal (SG) as input to a decorrelation module having a variable decorrelation rate. The decorrelation rate of the decorrelation module is modified as a function of the reception quality coefficient “alpha” provided by the radio. The decorrelation module applies a higher decorrelation rate for a smaller reception quality coefficient “alpha” and applies a lower decorrelation rate for a larger reception quality coefficient “alpha. Also, a module for generating high-pitched sounds to recreate the high-frequency component (SHF) of the right or left sound signals which has been removed in the event of poor reception.

Abstract: A mobile terminal includes an apparatus for reducing Time Division Multiple Access (TDMA) noise in the mobile communication terminal. The apparatus includes a first output path including a resistance element; a second output path not including a resistance element; a modem for, when detecting an event for outputting an audio signal, determining an output path of the audio signal and outputting a control signal indicating the determined output path; and a switch for switching according to the control signal and providing the audio signal to either the first output path or the second output path.

Abstract: A combined signal (Z) is provided as an additive mix of a secondary audio signal (Y) and a phase-inverted reduced primary signal (Xm?) obtained from a primary audio signal. The secondary signal (Y) can be restored from the primary (X) and the combined (Z) signal by additively mixing the latter with a reduced primary signal (Xm) obtained from the primary signal. This coding approach allows a supplementary audio service, in particular an audio description/video description to be distributed alongside with a multi-channel audio signal at low extra bandwidth or storage cost.

Abstract: A process and system for enhancing and customizing audio recording and editing comprises receiving an input audio sound; setting a preset for the input audio sound to create a presetted audio sound; tone adjusting the presetted audio sound to create a tone adjusted presetted audio; expanding the tone adjusted presetted audio in a horizontal plane to create an presetted tone adjusted expanded audio; and outputting the presetted tone adjusted expanded audio. There are three options for selecting the preset. These are: ten or more presets for different genres of music; auto-preset that is selected by genre in metadata of playback material; and a single generic preset that covers all music.

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

Abstract: This document describes techniques for synchronizing rendering of multi-channel audio output. First audio data corresponding to a first audio channel and second audio data corresponding to a second audio channel may be received. The first and second audio data may be configured for simultaneous rendering. A transmission schedule for wirelessly transmitting audio data to each of first and second audio rendering devices may be configured, according to which the first audio data may be scheduled for transmission to the first audio rendering device after the second audio data is scheduled for transmission to the second audio rendering device by a first length of time. An indication may be wirelessly transmitted to the second audio rendering device to delay rendering audio data by the first length of time. The first and second audio data may be wirelessly transmitted to the first and second audio rendering devices according to the transmission schedule.

Abstract: A master transmitter distributes a plurality of audio channels to one or more expansion receivers as a multiplexed audio stream. Control information is also transmitted between the master transmitter and the expansion receivers. Both the control information and the multiplexed audio stream are transmitted on the same cable allowing for reduced clutter and cheaper material and installation costs.

Abstract: The device for broadcasting sounds for playing an audio content coming from at least one audio content server includes: a base (220) for supplying electrical power to the device from a bulb socket, an electroacoustic transducer, a unit for discovering a network including at least the server and at least one other device for broadcasting sounds, and a unit for synchronizing the playing of sounds by the electroacoustic transducer and the playing of sounds by the other device. In embodiments, the device includes: a unit for exchanging messages with at least one other device for broadcasting sounds and a unit for selecting sound and/or light signals broadcast by the electroacoustic transducer, according to data received from at least one other device for broadcasting sounds. In embodiments, the selection unit is designed to select an audio channel according to at least one audio channel selected by another device for broadcasting sounds.

Abstract: Analog audio inputs are processed by a digital signal processor and rebroadcast over a local wireless network synchronous with a separate broadcast device or in real time with a live audio source. A software application on a portable computing device provides end-users the option to select from a plurality of audio source streams, cached translations of preprocessed audio streams and speech-to-text captioning. An embodiment of the invention relays wirelessly received digital audio streams converted from an analog source to the portable computing device then onto a hearing aid.

Abstract: An audio streaming system includes a handheld computing device that asynchronously streams audio to a storage subsystem. The handheld computing device receives and buffers audio data of contiguous audio samples and, after a predefined amount of contiguous audio samples has been buffered, assigns an index for that predefined amount of contiguous audio samples and asynchronously transmits the index and those contiguous audio samples to a storage subsystem of a broadcast server system. The handheld computing device, upon determining that the predefined amount of contiguous audio samples has been received by the storage subsystem, transmits a notification message to an application server of the broadcast server system that includes the assigned index and indicates that the corresponding contiguous audio samples are stored on the storage subsystem.

Abstract: A method, an apparatus, a computer readable storage medium configured with instructions for carrying out a method, and logic encoded in one or more computer-readable tangible medium to carry out actions. The method is to decode audio data that includes N.n channels to M.m decoded audio channels, including unpacking metadata and unpacking and decoding frequency domain exponent and mantissa data; determining transform coefficients from the unpacked and decoded frequency domain exponent and mantissa data; inverse transforming the frequency domain data; and in the case M<N, downmixing according to downmixing data, the downmixing carried out efficiently.

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: An audio video system is described that includes an AV receiver with a wireless audio module (WAM) host and without audio amplifier functionality. The system further includes a plurality of wireless speakers each having a WAM device to enable unidirectional or bidirectional communications with the WAM host. Each wireless speaker includes an amplifier that may be matched to a driver of the speaker to optimize the frequency response of the driver.

Abstract: This document discusses apparatus and methods for configuring and providing crosstalk cancellation to maintain channel separation in a multi channel system. In an example, an amplifier circuit can include a crosstalk cancellation circuit configured to reduce crosstalk from a first output to a second load and from a second output to a first load where the first load and the second load share a return path.

Abstract: System and methods are provided configured to reveal differences in content, such as commercial and program content. An input port may be configured to receive a first plurality of bit stream audio channels. A circuit may optionally be configured to enhance artifacting, noise, and/or bit rate errors and may optionally reduce redundant signal information. A signal processor circuit may be configured to transform the first plurality of bit stream audio channels into a frequency domain. A spectrum analysis system may be configured to compare properties of the first plurality of bit stream audio channels to one or more reference signals. An identification circuit may be configured to identify different content types included in the first plurality of bit stream audio channels.

Abstract: An audio encoder has a common preprocessing stage, an information sink based encoding branch such as spectral domain encoding branch, a information source based encoding branch such as an LPC-domain encoding branch and a switch for switching between these branches at inputs into these branches or outputs of these branches controlled by a decision stage. An audio decoder has a spectral domain decoding branch, an LPC-domain decoding branch, one or more switches for switching between the branches and a common post-processing stage for post-processing a time-domain audio signal for obtaining a post-processed audio signal.

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