Abstract: An audio signal processing apparatus performs audio signal process composed of a plurality of channels each having parameters used in the audio signal process. The audio signal processing apparatus has a plurality of channel strips, each channel strip being assigned with a channel and being provided with manipulators for adjusting values of the parameters of the assigned channel. There are provided a plurality of storing sections having different priorities relative to each other, each storing section being capable of storing a setting indicative of a channel set to a channel strip for assignment thereto. A changing section changes the settings stored in the storing sections.

Abstract: Provided is a capacitor-type sensor read-out circuit. The capacitor-type sensor read-out circuit includes: a signal conversion unit outputting a sensor signal inputted from a sensor; a voltage booster generating a bias voltage; and a capacitor-type signal coupling circuit receiving the sensor signal as a feedback, mixing the received sensor signal with the bias voltage, and outputting the mixed signal.

Abstract: A distributed self-scaling network audio processing system includes end nodes interconnected by packet-switched network and operating as peers on the network. Each of the end nodes supports local input processing, mixing, and output processing. The input processing includes the option of dual input channels for supporting separate front-of-house and monitor workflows. End nodes are added to the system to support specific audio processing applications, based on the number of audio sources, the number of output mixes required, and the number of locations from which users choose to interact with the system.

Abstract: An interface for expanding a signal starting from a first sensing signal and a second sensing signal, wherein a receiving intensity measuring element generates an intensity signal; and a selector is controlled to select each time the first sensing signal, the second sensing signal, or a combined signal deriving from a weighted combination of these signals. The selector uses a plurality of thresholds variable as a function of the intensity signal.

Abstract: Apparatus and method to mix first and second audio channels provided to a headset in response to the quantity of audio devices coupled to its connectors, in response to whether any of those audio devices provides more than one audio channel, and in response to their being a complete lack of electric power provided by any audio device and from a power source of the apparatus, itself.

Abstract: An acoustic coupling sensor 200 for a portable electronic device is described which includes an acoustic transmitter 22, a reference signal generator 20 coupled to the acoustic transmitter, and an acoustic sensor 24 wherein in operation, the reference signal generator is operable to transmit the reference signal to the acoustic transmitter, to detect an acoustic level of the transmitted reference signal via the acoustic sensor, and wherein the detected reference signal level corresponds to a value of acoustic coupling between the acoustic transmitter and the acoustic sensor. The acoustic coupling sensor may be used for example in a mobile phone to detect acoustic leakage in order to adapt the acoustic level in the mobile phone during use.

Abstract: A system and method for recording a video conference between a home location and at least one remote location. The method includes: receiving a first audio signal from a computer at the home location, the first audio signal including the audio feed of the video conference participant from the at least one remote location; feeding the first audio signal to (a) a sound producing device, and (b) a mixing device; receiving a second audio signal from a microphone at the home location, the second audio signal including an audio feed from the microphone; feeding the second audio signal to (a) an audio input of the computer at the home location and (b) the mixing device; combining, at the mixing device, the first and second audio signals into a third signal, wherein the third signal includes the audio feed from the microphone and the at least one remote location; outputting the third audio signal; and recording the outputted third audio signal on a recording device.

Abstract: Once a human operator selects one output channel using a port setting key and SEL key, output port parameters of a plurality of output ports connected with the selected output channel are displayed in a given arrangement on an output channel-port setting popup screen, and the thus-displayed output port parameters are allocated to controls of corresponding channel strips. Such arrangements allow parameter setting operation to be readily performed for each of the output ports.

Abstract: An audio control console includes a housing. The housing includes: an operating board provided with a plurality of rows of a plurality of channel strips each including a plurality of operating members each for controlling a value of a parameter of an audio signal processing in a signal processing channel; a support board comprising a flat blank region allowing an object to be placed thereon; and a stepped surface connecting the operating board and the support board to each other. The stepped surface connects the operating board and the support board to each other such that a height position of the support board is above a height position of the operating board. The blank region is inclined downward toward the stepped surface.

Abstract: A recording method includes acquiring each piece of acoustic data representing a sound from each of a plurality of portable terminal devices. The each of the plurality of portable terminal devices includes a recording unit configured to generate the piece of acoustic data. The recording method also includes executing synchronization processing for synchronizing the respective pieces of acoustic data and executing mixing processing for mixing a plurality of pieces of acoustic data for which the synchronization processing has been executed.

Abstract: A processing chip for a digital microphone and related input circuit and a digital microphone are described herein. In one aspect, the input circuit for a processing chip of a digital microphone includes: a PMOS transistor, a resistor, a current source, and a low-pass filter. The described processing chip possesses high anti high-frequency interference capabilities and the described input circuit possesses high high-frequency power supply rejection ratio.

Abstract: Signal processing section of a terminal converts acquired audio signals of a plurality of channels into frequency spectra set, calculates sound image positions corresponding to individual frequency components, and displays, on a display screen, the calculated sound image positions results by use of a coordinate system having coordinate axes of the frequency components and sound image positions. User-designated partial region of the coordinate system is set as a designated region and an amplitude-level adjusting amount is set for the designated region, so that the signal processing section adjusts amplitude levels of frequency components included in the frequency spectra and in the designated region, converts the adjusted frequency components into audio signals and outputs the converted audio signals.

Abstract: An I/O unit includes a plurality of ports, a mixer includes processing channels, and the mixer and the I/O unit are interconnected via a network. The mixer stores channel data per processing channel. The channel data includes virtual port data for indirectly controlling signal processing of a port to which the channel is patched. The I/O unit stores real port data for controlling a signal processing in a corresponding port per port. In response to a connection instruction, one port and processing channels are interconnected, and real port data of the one port connected with the processing channels is set to a value of virtual port data being maintained for the processing channels.

Abstract: In a digital mixer, when a patch of a certain channel is changed, values of parameters of one port which has been connected to the certain channel is applied as parameters of a newly connected port, thereby enabling to hand over parameters of the one port which have been set for the certain channel. Further, when a newly connected port is already connected to another channel, the digital mixer asks the user whether or not to apply the parameters of the one port.

Abstract: Embodiments described herein provide for acoustic signatures in a playback system. In some embodiments, a playback device includes an audio processing component configured to receive an audio input signal, receive an acoustic signature signal, generate an audio output signal based on the audio input signal and the acoustic signature signal, and output the generated audio output signal. In some embodiments, the acoustic signature identifies at least one of the playback device or a group of playback devices. Some embodiments additionally include the playback device amplifying the generated audio output signal with one or more amplifiers, and playing the generated audio output signal via one or more speakers.

Abstract: A channel conversion part (214) performs channel conversion on audio data of a plurality of channels so that the number of channels thereof is converted to an appropriate number of channels for which an acoustic effect can be verified or perceived by audience according to the volume level of reproduced sound, and audio is output only with a required number of channels. In addition, a frequency control part (215) for controlling the operating frequencies of switching regulators (220-223) of audio amplifiers (224-227) also performs frequency control according to the volume level of sound of each channel which performs an audio output.

Abstract: In a mixer system including an editing device that edits a configuration of signal processing by designating a combination of a plurality of components and a DSP that processes audio signals in accordance with the signal processing configuration edited by the editing device, when the mixer engine does not store a predetermined enabling key, audio signal processing including a protected component in the mixer engine is disabled, and when the mixer engine stores the predetermined enabling key, the audio signal processing is enabled. Further, the editing device is permitted to edit a signal processing configuration including the protected component irrespective of storage of the enabling key in the mixer engine.

Abstract: Apparatuses for and methods of carrying out dynamic equalization processing of an audio signal, and apparatuses for and methods of controlling such equalization processing of the audio signal to dynamically adjust the time-varying spectrum of an audio signal to more closely match a user specified target time-invariant perceived audio signal spectrum while preserving the original dynamic range of the audio signal. The dynamic equalization is according to a user-defined spectral profile specified by a control interface that allows a user to define, create, modify and/or apply the user-defined spectral profile.

Abstract: Provided are a method of generating and playing an object-based audio content that may effectively store preset information about an object-based audio content, and a computer-readable recording medium for storing data having a file format structure for an object-based audio service. The method of generating the object-based audio content may include: receiving a plurality of audio objects (310) generating at least one preset using the plurality of audio objects (320) and storing a preset parameter with respect to an attribute of the at least preset and the plurality of audio objects (330). The preset parameter may be stored in a form of a box that is defined in a media file format about the object-based audio content. Through this, it is possible to effectively store a preset about a plurality of audio objects.

Abstract: A system for transferring metadata between an audio mixing console and a digital audio workstation connected by an audio link and a computer network. The audio mixing console and the digital audio system implement an Ethernet-based communication protocol for sending messages between them over the network connection, the messages including metadata specifying a plurality of settings on the audio mixing console corresponding to audio information received by the audio mixing console during a live performance. The audio information is sent to the digital audio system over the high bandwidth connection. The high bandwidth connection can also be used to transfer the metadata. The transfer of metadata between the console and the digital audio workstation facilitates new digital audio archiving, virtual live mixing, and virtual soundcheck workflows.

Abstract: An audio system has a first channel for receiving a first input signal and driving a first speaker and a second channel for receiving a second input signal and driving a second speaker. A first feedforward circuit couples an input of the second channel circuit to an input of the first channel circuit. A second feedforward circuit couples an input of the first channel circuit to an input of the second channel circuit. Circuit parameters of the first and the second feedforward circuits are determined such that a first detected output signal is zero when the first input signal is non-zero and the second input signal is zero, and a second detected output signal is zero when the second input signal is non-zero and the first input signal is zero. The audio system is configured to operate using the determined circuit parameters for the first and the second feedforward circuits.

Abstract: A band broadening apparatus includes a processor configured to analyze a fundamental frequency based on an input signal bandlimited to a first band, generate a signal that includes a second band different from the first band based on the input signal, control a frequency response of the second band based on the fundamental frequency, reflect the frequency response of the second band on the signal that includes the second band and generate a frequency-response-adjusted signal that includes the second band, and synthesize the input signal and the frequency-response-adjusted signal.

Abstract: The present invention provides a foot-operated sound mixer, which, for example, allows a musical performer to adjust personal monitor mix by foot without using hands engaged in playing a musical instrument. The sound mixer of this invention has a housing that has at least one opening. The sound mixer of this invention also comprises an assembly of foot-controlled elements of a user interface comprising scroll wheel assembly and keys that are provided within the housing. The scroll wheel assembly includes a rotatable member positioned within the opening of the housing. The rotatable member is rotatable about an axis that extends within the housing and is laterally movable within the opening relative to the housing. The scroll wheel assembly also includes a movement sensing system that determines when the rotatable member is moved laterally relative to the housing.

Abstract: A personal audio device, such as a wireless telephone, includes noise canceling circuit that adaptively generates an anti-noise signal from a reference microphone signal and injects the anti-noise signal into the speaker or other transducer output to cause cancellation of ambient audio sounds. An error microphone may also be provided proximate the speaker to estimate an electro-acoustical path from the noise canceling circuit through the transducer. A processing circuit uses the reference and/or error microphone, optionally along with a microphone provided for capturing near-end speech, to determine whether one of the reference or error microphones is obstructed by comparing their received signal content and takes action to avoid generation of erroneous anti-noise.

Abstract: An apparatus comprising: at least one processor and at least one memory including computer code for one or more programs, the at least one memory and the computer code configured to with the at least one processor cause the apparatus to at least perform: receiving at least two signals comprising at least two audio signals from at least two recording apparatus recording within an audio scene an audio source, wherein the first of the at least two audio signals is configured to represent a first frequency range and the second of the at least two audio signals is configured to represent a second frequency range; scaling the at least two audio signals; and combining the at least two audio signals to generate a combined audio signal representation of the audio source.

Abstract: A multimedia control system which may be expanded without creating or modifying source code. The system is based on a master programmable multimedia controller. The master programmable multimedia controller has a list of unique identifiers that indicate valid subordinate programmable multimedia controllers that are eligible to become part of the system. One or more subordinate programmable multimedia controllers having a matching unique identifier to a unique identifier on the master's list may be added to the system to provide additional control ports, increase audio or video processing or switching capacity. A method is provided for determining whether a particular subordinate programmable multimedia controller may be added to the system.

Abstract: An audio media visualization method and system. The method includes receiving by a computing processor, mood description data describing different human emotions/moods. The computer processor an audio file comprising audio data and generates a mood descriptor file comprising portions of the audio data associated with specified descriptions of the mood description data. The computer processor receives a mood tag library file comprising mood tags mapped to animated and/or still objects representing various emotions/moods and associates each animated and/or still object with an associated description. The computer processor synchronizes the animated and/or still objects with the portions of said audio data and presents the animated and/or still objects synchronized with the portions of said audio data.

Abstract: Provided is an apparatus and method of encoding and decoding multiple channel signals based upon phase information and one or more residual signals.

Abstract: In order to realize a correction for changes in the reproduction loudness at low frequencies in a downmix-arrangement, a mixing arrangement is proposed for mixing at least two audio signals, which mixing arrangement is provided with a first unit (104) for deriving a first power signal, which is a measure for the power of the first audio signal, a second unit (105) for deriving a second power signal, which is a measure for the power of the second audio signal, a cross-correlation unit (103) for deriving a cross-correlation signal, which is a measure for a cross-correlation between the first and the second audio signal, a unit (106) for deriving multiplication parameters from the first and second power signals and the cross-correlation signal, and a multiplication and combination unit (107) for carrying out a signal processing on the first and second audio signals and combining them.

Abstract: Six input channels, to which are allocated 5.1-channel surround signals from a plurality of input sources, are grouped into a surround channel group, and these six input channels are connected to corresponding ones of six surround buses in a one-to-one relationship. Thus, merely grouping the input channels into a surround channel group allows the signals of the individual input channels (5.1-channel surround signals) to be taken out via a plurality of output destinations (5.1-channel speakers) corresponding to the surround buses. Once an instruction is given for changing a value of a parameter, the parameter is controlled in a ganged fashion in all of the input channels of the surround channel group except for an LFE input channel of the surround channel group.

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 multi-rate filter system is disclosed. More particularly, a computationally efficient multi-rate filter system for processing an audio stream on a consumer electronics device is disclosed. The multi-rate filter system includes a plurality of multi-rate filtering blocks, at least one block including a linear filter component. At least one multi-rate filtering block may include a nonlinear signal processing components. The multi-rate filter system may include a nonlinear functional block. A method of filtering a signal is also disclosed.

Abstract: Control data is generated in response to generation, from a position detector, of position data in accordance with a predetermined scale characteristic. Change scale characteristic is applied in manual operation performed immediately after an operation member has been automatically positioned to a target position. In the change scale characteristic, a start point is established such that control data, corresponding to the target position, of the predetermined scale characteristic is outputted in correspondence with the position data outputted at the time of completion of the positioning, but also a first change portion, including a portion extending from a position of the start point at least up to the target position, is established, so that the control data outputted in correspondence with the position data output gradually varies from data corresponding to the start point to data of the predetermined scale characteristic corresponding to an end point of the first change portion.

Abstract: An audio device (e.g. an AV amplifier), with a plurality of audio output terminals according to the HDMI standard, converts audio data of a source device into audio signals suited to audio reproduction abilities (e.g. channel counts) of sink devices. The audio device stores a plurality of audio reproduction ability information describing the predetermined channel count thereof and the channel counts of sink devices, wherein arbitrary audio reproduction ability information precluding audio reproduction ability information of a sink device with the smallest channel count is provided to a source device. The audio device performs down-mixing on audio signals with respect to a sink device whose channel count is smaller than the predetermined channel count, while the audio device performs signal processing on audio signals with respect to a sink device whose channel count is larger than the predetermined channel count.

Abstract: Methods, systems, and apparatus are provided for combining (e.g., mixing) audio signals received from a plurality of participants communicating with each other during a communication session (e.g., an audio conference) based on voice-activity-detection (VAD) data contained in extended headers of Real-time Transport Protocol (RTP) packets. An audio mixing apparatus receives RTP packets from connected clients and extracts the VAD data included in the extended RTP headers to render a mixing decision. Once a mixing decision has been made, the audio frames to be mixed are decoded while other received frames are discarded, thereby preventing processing resources from being wasted to decode frames that are never used.

Abstract: N input channel signals from an input channel section having N input channels are selectively supplied to M mixing buses and cue buses. Mixed outputs from the mixing buses are supplied, via talkback input sections, to an output channel section having M output channels, and M output channel signals from the output channel section are patched to output ports via an output patch section. Talkback signal from a talkback supply section is supplied to the talkback input sections. When a talkback function is ON, a talkback signal is mixed with the mixed outputs from the mixing buses and then output to the output channels, during which time the mixed outputs are temporarily attenuated.

Abstract: A system and a method for mixing at least two audio signals are provided that comprise transferring the audio signals with respective transfer functions, the audio signals each having an amplitude and a phase; adding the audio signals to provide an output signal representative of the mixed audio signals, the output signal having an amplitude and a phase; controlling at least one of the transfer functions of the signal lines so that the phase of the output signal is adapted to the phase of the audio signal with a higher signal strength than the other audio signal(s), the signal strengths corresponding to the amplitudes of the audio signals.

Abstract: A method of performing complementary mixing may include performing an exclusive OR (XOR) function with respect to an I-channel symbol based on an oscillator signal to produce an I-channel output signal with bits that alternate between the I-channel symbol and a complement of the I-channel symbol in response to the oscillator signal rising and falling. The method may also include performing the XOR function with respect to a Q-channel symbol based on the oscillator signal to produce a Q-channel output signal with bits that alternate between the Q-channel symbol and a complement of the Q-channel symbol in response to the oscillator signal. Further, the method may include combining the I-channel output signal and the Q-channel output signal based on adding operations performed with respect to an I-channel extra bit signal, a Q-channel extra bit signal, the I-channel output signal, and the Q-channel output signal to generate a complementary mixed signal.

Abstract: A device that enables easy, reliably performance of a search at a desired position even in the case of a one-bit digital signal. A control section of a digital signal processing device extracts a one-bit digital signal of a predetermined period from a memory section at a position designated by a jog dial and iteratively outputs the signal to a multi-bit converter. The multi-bit converter converts the one-bit digital signal into a multi-bit signal, and a multiplier subjects the multi-bit signal to fade-in and fade-out processing. A ?? modulator re-converts the thus-processed signal into the one-bit digital signal, outputting the digital signal.

Abstract: Systems and methods for mixing music are disclosed. Audio mix information is received from a plurality of users. Mix rules are determined from the audio mix information from the plurality of users, wherein the mix rules include a first mix rule associated with a first audio item. The first mix rule relates to an overlap of the first audio item with another audio item. The first mix rule is made available to one or more clients. In some implementations, making the first mix rule available to the one or more clients includes transmitting, to the first client, information enabling the first client to playback a transition between the first audio item and the second audio item in accordance with the first mix rule.

Abstract: A device (20) for controlling an audio signal comprising at least: means for commanding, rotationally mobile around an axis of rotation (A) relatively to a footing, first means for detecting a movement in rotation of the commanding means, capable of delivering a first signal, second means for detecting a movement in translation along an axis (z) substantially parallel to the said axis of rotation (A), capable of delivering a second signal. According to the invention, the said second means for detecting comprise at least one pressure sensor (210) delivering said second signal and comprising at least a upper part (210A), a lower part (210C), and a central part (210B) located between the upper part and the lower part, at least one of these parts being rigid, and at least one of these parts being elastic and deformable along an axis substantially parallel to the said axis of rotation (A).

Abstract: In a digital audio mixer and a method thereof, three effecter group lists corresponding to effecter group switches can be stored in a current memory. The first effecter group list can include member data identifying a plurality of effecters that constitute an effecter group. Each of the member data can include an effecter ID (eID) identifying an effecter (internal effecter, external effecter or default effecter) inserted in a desired insertion point of a desired channel. In response to operation of any one of the effecter group switches, a CPU can collectively set ON/OFF states of all of the effecters of the effecter group to a same state on the basis of the first effecter group list.

Abstract: Embodiments of the present invention include methods and apparatuses for adjusting audio content when more multiple audio objects are directed toward a single audio output device. The amplitude, white noise content, and frequencies can be adjusted to enhance overall sound quality or make content of certain audio objects more intelligible. Audio objects are classified by a class category, by which they are can be assigned class specific processing. Audio objects classes can also have a rank. The rank of an audio objects class is used to give priority to or apply specific processing to audio objects in the presence of other audio objects of different classes.

Abstract: A karaoke device is housed within an enclosure. A voice pickup element is integrated into the enclosure. The voice pickup element converts sound waves into an electrical signal. The karaoke device has an audio input and an audio output. An amplifier housed within the enclosure variably amplifies the electrical signal from the voice pickup element. A mixer, also within the enclosure, has a first input connected to an output of the amplifier and a second input connected to the audio input. A switch is include and has at least two positions. A first position of the switch connects the audio input to the audio output and a second position of the switch connects an output of the mixer to the audio output.

Abstract: A method for enhancing speech includes extracting a center channel of an audio signal, flattening the spectrum of the center channel, and mixing the flattened speech channel with the audio signal, thereby enhancing any speech in the audio signal. Also disclosed are a method for extracting a center channel of sound from an audio signal with multiple channels, a method for flattening the spectrum of an audio signal, and a method for detecting speech in an audio signal. Also disclosed is a speech enhancer that includes a center-channel extract, a spectral flattener, a speech-confidence generator, and a mixer for mixing the flattened speech channel with original audio signal proportionate to the confidence of having detected speech, thereby enhancing any speech in the audio signal.

Abstract: An audio mixer (20) include a chassis (22) adapted to receive a docking tray assembly (24), which in turn is adapted to receive a computing device (26). The computing device functions to operate and control the audio mixer (20), and to this end includes a touch screen display (28). The docking tray assembly (24) includes formed tray (110) and, optionally, an adapter configured as a formed insert (112) that nests with the tray (110) to dock a computing device of a configuration different from the computing device docked directly with the tray (110).

Abstract: An apparatus including an input configured to receive at least one first audio signal from a first apparatus; an input configured to receive at least one second audio signal from a second apparatus; a distance detector configured to determine a distance between the first and the second apparatus; and a level control generator configured to generate a level control for mixing the at least one first audio signal and the at least one second audio signal based on the distance between the first and the second apparatus.

Abstract: Systems and methods to deliver live sound with enhanced quality conveyed by a mixing desk to a mobile computing device user contemporaneously with external live sounds. The mobile computing device is operable to receive enhanced audio signals produced by a remote audio signal processing device through a communication work. A memory resident application is configured to playback the enhanced audio signals in phase with the external sounds using an audio rendering device. An attendee at the live event can hear the sounds from the playback through an earphone coupled to the mobile computing device as well as from the ambient environment.

Abstract: To a first analog signal that has been inputted, a first channel effect is added by an analog process that can be controlled with controls provided in a first channel strip. To a second analog signal that has been inputted, a second channel effect is added by a digital process, the second channel effect being controlled by a fourth control provided in a second channel strip. The second channel effect is preferably performed by a digital effect DSP.

Abstract: Mixer and one or more external devices, related to audio signal processing, are interconnected via a communication network, such as a LAN. For a particular one of the external devices for which communication with the mixer via the communication network is to be established, a network address unique to the particular external device and a communication protocol to be used for data communication with the particular external device are designated by entry operation by a user. Logical connection is established such that data communication is performed, in accordance with the designated communication protocol, between the mixer and the particular external device corresponding to the designated network address. Thus, data communication is performed, via the communication network, between the mixer and the particular external device for which the logical connection has been established.