Abstract: A system for controlling a vehicle sensor, the system including a plurality of sensors provided in a vehicle, a switch module including a plurality of switches connected to the plurality of sensors, respectively, the plurality of switches being connected to the switch module through an integral line; a controller configured to control operation of each of the plurality of switches, the controller being connected to the switch module through the integral line and thus performing communication with the plurality of sensors, and assigning an ID to each of the plurality of sensors in a corresponding manner by selectively connecting each of the plurality of switches thereto.

Abstract: In a video system for imaging a target object by using first and second imaging devices, an information processing method is executed by the first imaging device. The information processing method includes: a step for acquiring a first elapsed time indicating the time elapsed after start of recording by the first imaging device; a step for requesting the second imaging device to transmit a second elapsed time indicating the time elapsed after start of recording by the second imaging device, and receiving the second elapsed time; a step for acquiring a required time to receive the second elapsed time after the request is made; and a step for, based on the first elapsed time, the second elapsed time, and the required time, calculating an offset time for synchronizing a first moving image recorded by the first imaging device and a second moving image recorded by the second imaging device.

Abstract: A method and system for a passive headset with dynamically controlled LEDs, where the method comprises, in a passive headset comprising speakers, light emitting diodes (LEDs), and LED driver circuitry: receiving an electrical signal that includes an audio signal and an LED control signal, separating the audio signal and the LED control signal, communicating the audio signal to the speakers, communicating the LED control signal to the LED driver circuitry, and generating a bias voltage for each of the one or more LEDs utilizing the LED driver circuitry and the output LED control signal. A light output of the one or more LEDs may be configured utilizing the generated bias voltage. The amplifier may include a mixer that sums an audio signal with a control signal.

Abstract: A musical instrument special effects device, comprising a special effects unit connected by wire between the musical instrument and an output device, where the unit is operatively arranged to selectively produce at least one of a plurality of preprogrammed special audio effects; and a controller is operatively arranged to wirelessly control the special effects unit.

Abstract: A fader device according to an aspect of the present invention includes: a guide rail that extends linearly and a slider that slidably engages with the guide rail, in which the slider includes a pressure-contact member that is to be in pressure contact with the guide rail, and a compression coil spring for jutting the pressure-contact member toward the guide rail.

Abstract: Technologies for framework-level audio device virtualization include a computing device that executes multiple application framework instances. The computing device monitors for an application framework instance switch and, in response to an application framework instance switch, determines whether the current application framework instance is in the foreground. If in the foreground, the computing device selects a physical audio output device. The computing device may output audio data associated with the current application framework instance using a kernel audio driver associated with the physical audio output device. If not in the foreground, the computing device selects a null audio output device using a null audio hardware abstraction layer (HAL). The null audio HAL may sleep for the duration of audio data associated with the current application framework instance. The null audio HAL may be an operating-system- and device-independent shared library of the computing device.

Abstract: Methods and devices are provided for a wireless microphone network whereby robustness for reception of audio information transmitted by one or more wireless microphones is enhanced. The system incorporates a dual stage approach for collecting, transmitting and receiving audio information. In the first stage, audio information collected by one or more microphone modules is transmitted to a series of receiver base stations. In the second stage, information received at each base station is subsequently transmitted to a receiver hub that selects, blends and/or augments the information to produce a high quality representation for audio information that provides improved robustness and reliability with respect to the movement, physical placement or performance of each microphone module.

Abstract: Disclosed embodiments provide systems and methods related to measuring a frictionless audio-fader position. A frictionless audio-fader position measuring system comprises a control slider, a sensing magnet, at least two magnetic flux sensors configured to output at least one of a voltage or a current based on the position of the sensing magnet and, a processor wherein the processor calculates a correlation between the position of the sensing magnet and a phase angle between the outputs of at least two of the magnetic flux sensors to determine a position of the sensing magnet, wherein the determined position of the sensing magnet controls an amplifier, mixer, and/or other hardware or software component.

Abstract: An acoustic customization system to enhance a user's audio environment. One type of enhancement would allow a user to wear headphones and specify what ambient audio and source audio will be transmitted to the headphones. Added enhancements may include the display of an image representing the location of one or more audio sources referenced to a user, an audio source, or other location and/or the ability to select one or more of the sources and to record audio in the direction of the selected source(s). The system may take advantage of an ability to identify the location of an acoustic source or a directionally discriminating acoustic sensor, track an acoustic source, isolate acoustic signals based on location, source and/or nature of the acoustic signal, and identify an acoustic source. In addition, ultrasound may serve as an acoustic source and communication medium.

Abstract: A load-monitoring interfacing device obtains and processes sensor data from an electrical load. The interfacing device can include at least one power outlet to provide power to a corresponding electrical load. During operation, the interfacing device can obtain sensor data from a local power outlet coupled to an electrical load, such that the sensor data can indicate an electrical measurement associated with the electrical load. The interfacing device selects a rule to process based on the obtained sensor data, and processes the rule to determine whether to perform an action. If the rule's condition is satisfied, the interfacing device proceeds to processing the rule's action description to perform the action, such as to enable or disable a power outlet, or to perform any other pre-defined action.

Abstract: A switching device receives an output cable from a guitar. The output cable has a plurality of conductors corresponding to each of the pickups on the guitar, typically 2 or 3. A passthrough signal based on manually switched pickup selection corresponds to a main guitar output. A pickup buffer circuit on the guitar isolates the signal conductors from the selected passthrough signal, and also switches the pickup buffer circuit on and off to allow for backwards compatibility. A multi-conductor cable and jack simultaneously transmits the plurality of signals to the switching device. The switching device receives the multi-conductor cable. A combination circuit in the switching device combines the signals from the pickups and also delivers the individual pickup signals to a respective output jack. Each output jack simultaneously delivers the respective pickup or combined signal to a mixer, amplifier or other signal processing device for recording and/or further processing.

Abstract: Provided are a system and method of mixing a second audio stream with a first audio stream in an audio output device. The system is configured to execute the method, comprising buffering and outputting the first audio stream via the audio output device as unmodified output, determining at least one insertion spot within the first audio stream, modifying the first audio stream at an insertion spot to avoid content loss, outputting the second audio stream at the insertion spot, and resuming unmodified output of the first audio stream at or near a completion of the second audio stream. Modifying the first audio stream can include pausing and/or warping the first audio stream at the insertion spot. The audio output device can be a vehicle head unit or a wireless device, such as a mobile phone.

Abstract: In live music performances it is desirable for a mixing console operator to be able to hear the result of all the processing of a single input signal provided by the various processing steps without the other input signals being present. This is achievable by providing a console (10) which processes an input signal independently of other input signals to produce an operator output signal (210), wherein the processing is arranged to be identical to that applied in combination to the same signal as it is processed, in combination with other signals, for the creation of an output mix (180).

Abstract: The present disclosure relates to a file generation device, a file generation method, a file reproduction device, and a file reproduction method that enable reproduction of multi-channel liner PCM audio data in a file format. AudioSampleEntry is extended and a box specifying ORG LPCM (ORGLPCMSpecificBox) is assigned immediately thereafter. The box specifying ORG LPCM (ORGLPCMSpecificBox) is configured of fields, namely, from the top, audio data size (the number of bytes) (audio_data_payload_size), channel assignment (channel_assignment), sampling frequency (sampling_frequency), number of bits of the sample (quantization bitrate) (bits_per_sample), and reservation area (reserved). The present disclosure can be applied to a signal processing system including a file generation device and a file reproduction device, for example.

Abstract: The present disclosure relates to user interfaces for manipulating user interface objects. A device, including a display and a rotatable input mechanism, is described in relation to manipulating user interface objects. In some examples, the manipulation of the object is a scroll, zoom, or rotate of the object. In other examples, objects are selected in accordance with simulated magnetic properties.

Abstract: There is provided systems and methods for storing and transmitting dynamic range tags in video using a non-transitory memory storing a video having a plurality of frames, a plurality of dynamic range tags corresponding to the plurality of frames, and a video tagging software, and a hardware processor executing the video tagging software stored in the non-transitory memory to obtain a frame of the plurality of frames of the video from the memory, obtain a dynamic range tag including information about the frame of the plurality of frames of the video from the memory, insert the dynamic range tag into a vertical ancillary data (VANC) space of the frame of the video to generate a modified video.

Abstract: In one embodiment, a polarity optimizer determines optimal polarities of audio input channels. In operation, the polarity optimizer generates multiple polarity combinations, where each polarity combination specifies a different combination of polarities of the audio input channels. For each of these polarity combinations, the polarity optimizer combines inverted and/or non-inverted samples of the audio input channels to generate a corresponding audio mix. For each of the audio mixes, the polarity optimizer calculates the value of a pre-determined signal characteristic. The polarity optimizer then compares these values and sets the optimized polarities in accordance with the polarity combination corresponding to the audio mix with the optimal value.

Abstract: An audio mixing system comprises a plurality of input ports and output ports, and a plurality of input channels and output channels for controlling characteristic of input sound signals. Two or more input ports included in the plurality of input ports or two or more output ports included in the plurality of output ports are registered as a plurality of patch ports. By an instruction to collectively patch a port group, the input ports or the output ports registered as the plurality of patch ports are patched to input channels or output channels respectively by an input or an output patch.

Abstract: A sound system and method for capturing and reproducing sounds produced by a plurality of sound sources are disclosed. Sounds produced by the plurality of sound sources may be received and converted separately to separate audio signals without mixing the audio signals. The audio signals may be separately stored on a recording medium without mixing the audio signals. Such audio signals may be read from the recording medium and separately amplified using an amplification network of amplifiers. Loudspeakers in a loudspeaker network may be used to separately reproduce the sounds from the amplified audio signals. Components within a signal path may be dynamically controlled by a controller. The amplifiers and/or loudspeakers for each signal path may be customized based on the characteristics and complexities of the original sound to be reproduced on each signal path.

Abstract: An apparatus including an audio mixer configured to mix at least two audio channels of an audio signal; and a sensor configured to sense motion and/or orientation of the apparatus. The audio mixer is configured to change mixing of the at least two audio channels based, at least partially, on an output from the sensor regarding the sensed motion and/or orientation of the apparatus.

Abstract: A mixing board for sound signals received on at least two channels, which can include the following circuits: a circuit for pre-listening to the sound signals through headphones that can be connected, a circuit for changing the timbre by means of a sound filter, and a circuit for cross-fading and/or mixing sound signals of the channels by means of a cross-fader, and which comprises several outputs, wherein the sound signals received on the channels are headphone output signals of players, the relevant channels are divided by means of circuitry and the signals are passed on to the outputs in parallel and without amplification: a) through a passive sound filter and subsequently a passive cross-fader to an output to which an amplifier can be connected, and b) via a switch (6, 7) of the relevant channel to a headphone output of the mixing board, wherein the channels can be selected by means of the switches.

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: A multitrack recorder capable of setting one track or more to a monaural type or a stereo type has a storage unit that stores a history of track-type changing-and-setting operations and operations entailing erasure or changing of audio data stored on each track and that stores specifics of the track-type changing-and-setting operations and the audio data erased or changed along with performance of operations in association with each operation, and a change unit that changes audio data stored in each track, and a type of the track, to audio data that were stored in each track, and a type of the track, before performance of a specified operation.

Abstract: Method and system for capturing and playing back ancillary data associated with a video stream. At capture, a first video stream and its associated non-audio ancillary data are received. The non-audio ancillary data associated with the first video stream is encoded into a first audio stream on a basis of a predefined encoding scheme. The captured non-audio ancillary data can then be transmitted and processed with the first video stream in the form of the first audio stream. At playback, a second video stream and a second audio stream containing encoded non-audio ancillary data associated with the second video stream are received. The second audio stream is decoded on a basis of a predefined decoding scheme in order to extract therefrom the non-audio ancillary data associated with the second video stream. The second video stream and its associated non-audio ancillary data are then both output for playback.

Abstract: A multitrack recorder can allocate an input source in a simple manner without regard to whether a track is a monaural type or a stereo type. The multitrack recorder can set tracks 3 and 4, among a plurality of tracks 1 through 4, to either a monaural type or a stereotype. Track types are stored in memory. On occasion of allocation of an input source to each of the tracks, available input source alternatives are displayed in a display section while being changed according to whether the track is a monaural type or a stereo type.

Abstract: In a RAM 1-3, a data area and a current memory are provided. A signal processing portion 1-10 processes audio signals in accordance with a parameter set stored in the current memory. In the data area, a plurality of S data sets each formed of a parameter set and an S list which is a list of S data sets are stored. Each S data set is identified by its unique identification information (SID). The S list stores list information including SIDs which identify the respective S data sets. When storage of the parameter set stored in the current memory is desired, a new SID is generated, so that list information including the generated SID is added to the S list, with the SID being added to the parameter set of the current memory to be stored as a new S data set in the data area.

Abstract: An audio locating and tracking apparatus, a method of directing a camera to view an audio source, and a video conferencing terminal are disclosed herein. In one embodiment, the apparatus includes: (1) an audio source identifier configured to locate an audio source based on multimodal sensor data from at least two different types of sensors and (2) an image selector configured to automatically direct a camera to view the audio source.

Abstract: An apparatus for controlling a wave field synthesis renderer with audio objects includes a provider for providing a scene description, wherein the scene description defines a temporal sequence of audio objects in an audio scene and further includes information on the source position of a virtual source as well as on a start or an end of the virtual source. Furthermore, the audio object includes at least a reference to an audio file associated with the virtual source. The audio objects are processed by a processor, in order to generate a single output data stream for each renderer module, wherein both information on the position of the virtual source and the audio file itself are included in mutual association in this output data stream. With this, high portability on the one hand and high quality due to secure data consistency on the other hand are achieved.

Abstract: A method is designed for controlling a total mixing system including a first mixing system and a second mixing system, which are operated in a linked manner. In the method, the first mixing system stores first scene data specifying contents of a mixing process matching a scene. The second mixing system stores second scene data specifying contents of a mixing process matching a scene. The first mixing system transmits a scene recall request to the second mixing system when a recall event of the first scene data occurs. The second mixing system transmits back a recall enabling response to the first mixing system after receipt of the scene recall request. The first mixing system reconstructs the contents of the mixing process on the basis of the first scene data after the reception of the recall enabling response. The second mixing system reconstructs the contents of the mixing process on the basis of the second scene data after the transmission of the recall enabling response.

Abstract: A system and method is introduced to facilitate the handover process of performers who are using digital vinyl systems (DVS). A DVS may be connected to an existing audio setup during the playing of another performer without interrupting the other performer and without disconnecting any connections in the existing audio setup. When the DVS is connected and a handover process is required, a switch is activated to select the DVS as the current active device for use. A DVS may be disconnected from an existing audio setup without interrupting another performer. The system may be implemented as part of an audio device or as a separate stand alone device.

Abstract: There is provided a recording apparatus which can effectively execute compression processing regardless of the mixdown level, and which can thereby carry out the mastering process easily. A DSP mixes down audio signals assigned to a plurality of tracks into a stereo audio signal. When performing the mastering process, the DSP uses a peak value detected during the mixdown to adjust a threshold value for the compression processing, and carries out the compression processing using the adjusted threshold value. Subsequently, the DSP performs normalization processing so as to produce a master data, and records the master data in a recorder.

Abstract: Main and sub assignable controls are provided for each channel strip. Predetermined screens, each including a parameter display, are displayed dividedly or separately on first and second display devices. Thus, any one of a plurality of parameters included in the predetermined screen, being displayed on the first display device, is variably assigned to the main assignable control, while any one of a plurality of parameters included in the predetermined screen, being displayed on the second display device, is variably assigned to the sub assignable control. Further, when a predetermined pop-up screen, including at least a parameter display, is to be further displayed, in response to a screen display switching instruction, over the predetermined screen being displayed on the first display device, a parameter included in the pop-up screen is fixedly assigned to each of the main and sub assignable controls. In this way, the plurality of assignable controls can be used properly in accordance with a screen display.

Abstract: A method of processing an audio signal, comprising: receiving a downmix signal, a residual signal and object information; extracting at least one of a background-object signal and a foreground-object signal from the downmix signal using the residual signal; receiving mix information comprising gain control information for the background-object signal; generating a downmix processing information based on the object information and the mix information; and, generating a processed downmix signal comprising a modified background-object signal to which an adjusted gain corresponding to the gain control information is applied, by applying the downmix processing information to the at least one of the background-object signal and the foreground-object signal is disclosed.

Abstract: A multitrack recorder capable of setting one track or more to a monaural type or a stereo type has a storage unit that stores a history of track-type changing-and-setting operations and operations entailing erasure or changing of audio data stored on each track and that stores specifics of the track-type changing-and-setting operations and the audio data erased or changed along with performance of operations in association with each operation, and a change unit that changes audio data stored in each track, and a type of the track, to audio data that were stored in each track, and a type of the track, before performance of a specified operation.

Abstract: An audio recording software program adapted to capture spontaneous extemporaneous performance after group recording. The software permits a user to suspend recording of some tracks while continuing recording of at least one track, to capture extemporaneous performances of musical phrases, especially vocal performances, which are often preferred in the extemporaneous utterance which are difficult to recapture.

Abstract: A system, device, and method for recording audio with the character and sonic benefits of a genuine analog recording is disclosed. More specifically, an electro-mechanical-software controlled closed loop analog signal processor (“CLASP”) system which is comprised of a CLASP unit containing firmware, a latency detection module, and CLASP hardware display and controls. The CLASP system further comprises CLASP software operably running on a digital audio workstation (“DAW”) which is also in operable communication with the CLASP unit. The CLASP unit is also in operable communication with an analog recordable medium. An analog audio signal is recorded on the analog recordable medium, which may consist of a coated tape, cup, cylinder, drum, or disk, and then immediately played back and routed to the DAW via an analog to digital converter, thus providing for digitally recorded analog audio. The CLASP system may also include converters and a mixing console.

Abstract: An apparatus for processing an audio signal and method thereof, comprising receiving a downmix signal comprising at least one normal object signal, and bitstream including object information determined when the downmix signal is generated; extracting extension type identifier indicating whether the downmix signal further comprises a multi-channel object signal, from extension part of the bitstream; when the extension type identifier indicates that the downmix signal further comprise multi-channel object signal, extracting first spatial information from the bitstream; and, transmitting at least one of the first spatial information and the second spatial information; wherein the first spatial information is determined when a multi-channel source signal are downmixed into the multi-channel object signal, wherein the second information is generated using the object information and mix information, are disclosed.

Abstract: An apparatus for signal processing, wherein a disc is placed on a turntable and is provided with a groove which can be followed by the pick-up element, and employing a time-code signal wherein during use of the disc the said time-code signal controls the digital audio source.

Abstract: An apparatus for processing an audio signal and method thereof are disclosed.

Abstract: A device for audio mixing includes a cascade input port and a first cascade input component that includes a first cascade input level detector component that detects an audio level of an upstream sum audio signal. The first cascade input port also includes an attenuator component that attenuates the first upstream mix audio signal by a gain corresponding to difference between the upstream sum audio signal value and a detected audio level of an input sum audio signal. The device also includes a summer component where the signals summed include at least the first upstream sum audio signal. The device also includes an input sum level detector component that detects the audio level of the input sum audio signal. The device also includes a mixer component that is configured to provide a mix output signal by summing, where the signals summed include at least the first attenuated upstream mix signal.

Abstract: A method of processing a signal is disclosed, which includes receiving a downmix signal, object information and rendering information; obtaining downmix gain flag information indicating whether a downmix gain is included in the object information; and generating downmix processing information using the object information and the rendering information based on the downmix gain flag information, wherein the object information includes modified object level information or object level information, wherein the modified object level information is determined using a level of a modified object generated by applying the downmix gain to an object, and wherein the object level information is determined using a level of the object. Accordingly, in an apparatus for processing a signal and method thereof according to the present invention, the signal is decoded using modified object level information based on downmix gain flag information.

Abstract: Content reproducing apparatus and method of a mobile terminal are disclosed. During the reproduction of content, vibration pattern corresponding to content is detect or generated. The vibration pattern is synchronized with an audio signal during the reproduction of content.

Abstract: A method of automatic announcer voice removal from a televised sporting event. A sound processing circuit divides an audio input signal of a televised sporting event into multiple audio segments. The audio input signal includes crowd noise and announcer commentary. If an audio segment does not exceed a pre-defined amplitude threshold, a voice removal utility adds the audio segment to a recent crowd noise library and stores the segment in an output buffer. If the amplitude of a segment exceeds the threshold, the utility adds the segment to a recent announcer voice library. The sound processing circuit generates an attenuated version of the segment and blends the attenuated version with one or more mixed segments from the recent crowd noise library. The voice removal utility stores the attenuated and blended segment in the output buffer and outputs one or more audio segments from the buffer in a chronological order.

Abstract: A digital mixer is configured to store, in response to a store operation by a user, values of parameters for one input channel stored in a current memory together with the bus type of each mixing bus at the time of the store into a library as one preset, and to recall, when recalling the values of the parameters in the stored one preset into the current memory in response to a recall operation by the user, the values of the parameters in the preset after converted into parameters corresponding to the bus type at the time of the recall regarding a part of the buses whose bus type stored does not coincide with the bus type at the time of the recall.

Abstract: A user-friendly system for real time performance and user modification of one or more previously recorded musical compositions facilitates user involvement in the creative process of a new composition that reflects the user's personal style and musical tastes. Such a system may be implemented in a small portable electronic device such as a handheld smartphone that includes a stored library of musical material including original and alternative versions of each of several different components of a common original musical composition, and a graphic user interface that allows the user to select at different times while that original composition is being performed, which versions of which components are to be incorporated to thereby create in real time a new performance that includes elements of the original performance, preferably enhanced at various times with user selected digital sound effects including stuttering and filtering.

Abstract: A distributed audio signal processing system having a plurality of linked audio signal processing units is disclosed. Each audio signal processing unit has physical channels for receiving and sending local audio signals and a high bandwidth interface for exchanging audio signals with other linked audio signal processing units. Each of the physical channels of each of the audio signal processing units are mapped to a corresponding global channel. The audio signal processing system provides a tightly integrated, distributed signal processing matrix where any input signal or physical channel on any one of the signal processing units can be exchanged with any other of the linked audio signal processing units, thereby providing maximum flexibility for selecting signals to mix and determining how to process the signals.

Abstract: Described herein are one or more implementations offering a centralized (e.g., single-point) user-interface, where a user may manipulate audio properties (e.g., volume level) for one or more audio submixes produced by each of multiple active operating-system processes.

Abstract: A method and apparatus for multitrack recording of audio information using handheld digital electronic devices are described. Recording of video tracks complementing the audio is also contemplated. One device serves as master; others, slaves. Transmission of start and stop recording signals by the master may facilitate synchronization of the tracks. Slaves may begin preliminary sampling of audio information prior to the start signal, preparing to promptly begin actual track recording. Upon receiving the command to start recording, each slave chooses as first frame that sample closest to the signal to record. The task of stopping recording is typically handled similarly. A recording device may store an indicator for determining the offset of its first frame, facilitating time interpolation of samples in its track. In this way, all audio tracks, for example, might be transformed to a common set of times. The master might also record a track.

Abstract: A distributed audio signal processing system having a plurality of linked audio signal processing units is disclosed. Each audio signal processing unit has physical channels for receiving and sending local audio signals and a high bandwidth interface for exchanging audio signals with other linked audio signal processing units. Each of the physical channels of each of the audio signal processing units is mapped to a corresponding global channel. Global channels can be combined to form virtual channels that can be processed as a signal channel.

Abstract: Tools and techniques are provided to allow the user of a signal editing application to retain control over individual changes, while still relieving the user of the responsibility of manually identifying problems. Specifically, tools and techniques are provided which separate the automated finding of potential problems from the automated correction of those problems. Thus, editing is performed in two phases, referred to herein as the “analysis” phase and the “action” phase. During the analysis phase, the signal editing application automatically identifies target areas within the signal that may be of particular interest to the user. During the “action” phase, the user is presented with the results of the analysis phase, and is able to decide what action to take relative to each target area.