Abstract: There are disclosed automatic mixers and methods for creating a surround audio mix. A set of rules may be stored in a rule base. A rule engine may select a subset of the set of rules based, at least in part, on metadata associated with a plurality of stems. A mixing matrix may mix the plurality of stems in accordance with the selected subset of rules to provide three or more output channels.

Abstract: Provided is a decoding device including a reception unit that receives data of which the number of bits is converted and encoded such that a ratio between appearance frequency of a first code and appearance frequency of a second code is a predetermined range, and to which an error correcting code including redundant bits for calculating an error position of the data and a parity check bit of the data is appended, and a detection unit that detects that there are an odd number of bit errors in the data when a value of a syndrome corresponding to an error position is a first predetermined value and an error occurs in the decoding, or when the value of the syndrome is not the first value and a value of the parity check bit is a second predetermined value and an error occurs in the decoding on the data.

Abstract: An audio signal processor receives a plurality of encoded multi-channel audio signals. A multi-channel decoder (105) decodes a first encoded multi-channel signal to generate a first decoded multi-channel signal. A generator (109) generates an encoded further audio signal by selecting audio encoding data from at least a second encoded multi-channel audio signal such that a number of channels of the encoded further audio signal comprising audio encoding data from the second encoded multi-channel audio signal is less than a number of channels in the second encoded multi-channel signal. Thus, a channel reduction is performed in the encoded data domain. A further decoder (111) generates a further decoded signal by decoding the further encoded audio signal. A combiner (107) combines the first decoded multi-channel signal and the further decoded signal to generate a multi-channel output signal. An exciting user experience can be provided while maintaining low complexity and resource usage.

Abstract: While level of the inputted analog signal is adjusted according to a set analog gain value and then analog-digital converted and outputted, when an automatic compensation is on and a difference of the analog gain value and a target value is compensated by level adjustment in a digital amplifier, the analog gain value is displayed by a gain knob, and in response to an operation to turn on the automatic compensation, a present value of the analog gain value at that time is displayed as a target gain value by a mark near the analog gain value, and in response to an operation to turn off the automatic compensation, display of the target gain value is erased.

Abstract: When an effect parameter of an effector is changed, if only a switching operation is provided to bypass the original audio signals, which are input signals of the effector, during the period required for making that change, the output sound level may drastically change before and after the switching causing one to feel a sense of discomfort. In order to eliminate such a drawback, a gain adjustment circuit is provided in a bypass channel that outputs the original audio signals by bypassing the effector. The gain adjustment circuit is controlled in accordance with a rate of the output signal level of the effector and the original audio signal level, and the output signal level of the bypass channel is adjusted thereby. Consequently, a fluctuation in the output sound level before and after switching of the output signals can be eliminated.

Abstract: An audio signal processing apparatus acquires a specific audio signal based on at least one of audio signals of multiple sound capturing channels captured by multiple microphones, cancels a signal corresponding to the specific audio signal in the audio signals of the sound capturing channels that are to be used in surround audio signal generation, and, using the specific audio signal and the audio signals in which the signal corresponding to the specific audio signal was canceled, generates a surround audio signal such that the sound source of the specific audio signal is located at a specified direction in the surround sound field.

Abstract: A signal processing circuit comprising: a first signal processor configured to produce a first output signal suitable for driving a first headphone speaker; a second signal processor configured to produce a second output signal suitable for driving a second headphone speaker; a third signal processor configured to receive the first input signal and/or the second input signal and to produce a third output signal suitable for driving a first loudspeaker; and simultaneously provide the first output signal to the first headphone speaker, the second output signal to the second headphone speaker, and the third output signal to the first loudspeaker.

Abstract: Systems and methods are disclosed relating to electro-mechanical devices and related computer control and audio processing systems intended to optimize playback fidelity of phonograph records. Said recordings are manufactured by a process that employs a cutting head assembly driven in a straight path from the outer to the inner-most radius of the recordable surface. However, most record turntables device that are used to play back phonograph records rely on a stylus transducer attached to the end of a pivoting arm. Instead of the linear path followed by the original cutting head, said tonearm traces an arc across the surface of the recorded disk resulting in playback distortion proportional to error in alignment of said stylus relative to the tangent of the groove. This invention addresses this deficiency and produces an optimal audio quality of playback of phonograph records.

Abstract: A system for multi-track playback of media content includes: a media device; a user interface, provided at the media device, which displays a visual array of media options, a playback logic, provided within the media device, which is configured so that, while a selected point or region is determined by the user interface as being moved in response to user input, within the visual array of media options, the system determines media options that are proximate to the selected point or region, and adjusts playback parameters for corresponding media content items, by crossfading or otherwise combining playback to reflect the media options relative distances from the selected point or region; and a tempo logic, provided within the media device, which is configured to provide or receive a selected tempo and provide the one or more media content items associated with the selected tempo.

Abstract: An audio object coder for generating an encoded object signal using a plurality of audio objects includes a downmix information generator for generating downmix information indicating a distribution of the plurality of audio objects into at least two downmix channels, an audio object parameter generator for generating object parameters for the audio objects, and an output interface for generating the imported audio output signal using the downmix information and the object parameters. An audio synthesizer uses the downmix information for generating output data usable for creating a plurality of output channels of the predefined audio output configuration.

Abstract: An in-car communication (ICC) system has multiple acoustic zones having varying acoustic environments. At least one input microphone within at least one acoustic zone develops a corresponding microphone signal from one or more system users. At least one loudspeaker within at least one acoustic zone provides acoustic audio to the system users. A wind noise module makes a determination of when wind noise is present in the microphone signal and modifies the microphone signal based on the determination.

Abstract: The invention relates to an operating device (1) for operating at least one electric device (2) that has one, two, or more functional parameter(s) (3), said operating device (1) comprising a touchscreen (4) and a mechanical operating element (5). The operating device (1) is designed to assign control over the functional parameter(s) (3) to the mechanical operating element (5) by means of a selection on the touchscreen (4).

Abstract: There is provided a signal processing apparatus including a provision control unit that provides choices expressed subjectively and associated with control for an acoustic signal, and an output unit that outputs a predetermined choice selected from the provided choices.

Abstract: A universal effects carrier for use with electric instruments that allows for reduced or eliminated signal loss before the electronic signal of the instrument is passed through an effects pedal. A mounting cartridge includes a first receiving slot and a second receiving slot for retaining a primary effects pedal and a secondary effects pedal; the primary effects pedal and the secondary effects pedal being electrically connected to a signal-in port and a signal-out port via a signal-in pedal connector and a signal-out pedal connector. The primary effects pedal and the secondary effects pedal are further electrically connected by a bridge cable. The mounting cartridge is attached to an electric instrument through a cartridge receiver; the cartridge receiver including a signal-in terminal to be electrically connected to the signal-in port and a signal-out terminal to be electrically connected to the signal-out port.

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 apparatus for processing an audio signal and method thereof are disclosed. The present invention includes receiving a downmix signal and side information; extracting control restriction information from the side information; receiving control information for controlling gain or panning at least one object signal; generating at least one of first multi-channel information and first downmix processing information based on the control information and object information, without using the control restriction information; and, generating an output signal by applying the at least one of the first multichannel information and the first downmix processing information to the downmix signal, wherein the control restriction information relates to a parameter indicating limiting degree of the control information.

Abstract: Systems and methods presented herein may provide for processing audio with increased loudness and dynamics. A first clock frequency associated with a digital audio file may be increased to a second clock frequency, which speeds up the playback of the audio. The digital audio file may then be converted to analog at the higher second frequency, where it is further modified in the analog domain. The analog audio is then converted back into digital audio at the second clock frequency. The clock frequency is then decreased back to the first clock frequency for playback or storage of the processed (i.e., modified or manipulated) digital audio file. Alternatively, the entire process can take place in the digital domain.

Abstract: A system and method of determining the appropriate mixing volume corresponding to an impact related event is described herein. The instant invention facilitates determination of the appropriate mixing volume for the event sound corresponding to an impact related event in an related audio-video feed so as to account for the perceived visual proximity of the event and variations in the loudness of accompanying event audio, thereby enabling perfect mixing of event audio and event sound. The instant invention further discloses method/s of minimizing any unwarranted/drastic increase in the overall volume during the mixing process.

Abstract: A method includes steps determining individual ones of speakers and microphones connected to a first computerized appliance by execution of a software routine, determining individual ones of computer applications executable on the first or a second computerized appliance, and capable of audio input and output, and associating the speakers and microphones with the computer applications such that audio output from individual applications is provided to associated speakers only and audio input from individual microphones is provided to associated applications only.

Abstract: It is determined whether user's operation performed on a control is normal or special operation. A function of increasing or decreasing a value of a parameter assigned to the control is performed if the user's operation is the normal operation. Another function, different from the function of increasing or decreasing a value of a parameter, is performed if the user's operation is the special operation. The control comprises, for example, a touch pad provided with a display device for displaying a current operating position of the control. If a user touches one point of the pad with its finger or touches one point of the pad and then continuously slides its finger on and along the pad, such user's operation is determined to be the normal operation. If the user touches or taps two or more points of the pad, such user's operation is determined to be the special operation.

Abstract: A method and system including segmenting digital samples related to input audio data into arc segments representing output audio data. The arc segments are arranged to form multiple arcs which are arranged to form a record image. A user-selected boundary of a track of the input audio data may be marked based on a command received from a user via an interaction with the record image.

Abstract: Methods and systems are provided for independent audio volume control. For example, an audio component (e.g., implemented in an element of an audio setup) may control generation and/or outputting of a combined-game-and-chat audio signals. The audio component may receive user input comprising one or both of: a setting for a desired volume of a chat audio component of the combined-game-and-chat audio signal, and a setting for a desired volume of a game audio component of the combined-game-and-chat audio signal. The audio component may determine, based on the user input, a corresponding mix setting for mixing together of the chat audio component and the game audio component in an audio source to generate the combined-game-and-chat audio signal and a corresponding volume setting applicable at an audio output element; provide the mix setting to the audio source; and provide the volume setting to the audio output element.

Abstract: A digital mixer includes a MO switch and a MO fader in a signal transmission route from each input channel to a bus of a MIX_MINUS type, a signal of the input channel is supplied to the MIX bus without control on its level when the MO switch is OFF, while being supplied to the MIX bus after being controlled on its level by the MOL fader when the MO switch is ON. When the MO switch is changed from OFF to ON, a level adjustment amount by the MOL fader is set to a value so that a signal passing therethrough comes to have zero level.

Abstract: A cartridge receiver for a tone effects system used with an electric instrument that allows for reduced or eliminated signal loss before sound effects are applied. The cartridge receiver is integrated into the body of a desired electric instrument, while an effects cartridge is attached to the electric instrument via the cartridge receiver; the cartridge receiver providing electronic connections between the electric instrument and the effects cartridge. The effects cartridge provides an effects circuit for manipulating the received electronic signal, as well as an at least one effects control for adjusting the extent to which the effect is applied to the electronic signal. The cartridge receiver can be internally mounted within the electric instrument, wherein the cartridge receiver is fully recessed, or the cartridge receiver can be externally mounted to the electric instrument.

Abstract: A method, medium, and system decoding and/or encoding multiple channels. Accordingly, down-mixed multiple channels can be decoded/up-mixed to a left channel and a right channel during a first stage, thereby enabling a high quality sound output even in scalable channel decoding.

Abstract: A controller includes a plurality of channel strips to which are allocated any one of a plurality of channels to be handled in a DAW. Each of the channel strips includes a channel strip display section, a movable fader capable of indicating a sound volume level of the allocated channel by means of a knob position of the fader, and a slide operation reception section operable to successively change the allocated channel. During a slide operation on the reception section, a CPU continuously updates only displayed content on the channel strip display section of each of the channel strips by a variation amount corresponding to the slide operation. Upon completion of the slide operation on the reception section, the CPU moves, through automatic control, the knob position of the movable fader of each of the channel strips.

Abstract: A surround audio dialog balance assessment method, apparatus, and system as disclosed herein is an audio monitor or is associated with an audio monitor. Preferred surround audio dialog balance assessment methods, apparatuses, and systems automate the process of monitoring audio signals through a broadcast chain.

Abstract: A system for improving intelligibility or loudness of an audio program includes an encoder that receives signals of the audio program including at least one of left/front and right/front or left and right signals that include some anchor components of the audio program and to downmix the received signals to obtain left downmix and right downmix signals. The system includes a decoder that upmixes the left downmix and right downmix signals to obtain a center upmix signal that includes a majority of the anchor components including at least some anchor components that were included in the left/front and right/front signals or the left and right signals. The system also includes a system output that provides the center upmix signal to process at least one of the signals of the audio program based on the center upmix signal to improve intelligibility or loudness of the audio program.

Abstract: As viewed from the front of an operation panel, adjustment sections for individual input channels are positioned on the operation panel close to a human operator, and a display, a control section for setting an equalizer process, compressor process etc. for one selected channel and a signal delivery control section for setting signal delivery to mixing buses for the selected channel are positioned in a horizontal row remotely from the human operator. Operators in the delivery control section are varied in function in accordance with a selected operation mode. For example, in a mix send mode, a plurality of rotary operators are set to function as send level adjustment operators for the respective input channels so as to adjust send levels to a given mixing bus, while, in a mix master mode, the rotary operators are set to function as output gain adjustment operators for the respective mixing buses.

Abstract: An output channel processing an audio signal after mixing in a mixing bus in a digital mixer is configured such that output points PreHPF and PostON are provided at locations before and after a signal processing module group composed of signal processing modules from a high-pass filter to an ON/OFF control module, an output selecting switch selects one of the output points to supply the audio signal at the selected output point to a direct-out output module which is provided corresponding to the output channel and includes an ON/OFF control module and a level adjustment module so that the audio signal can be outputted via the direct-out output module.

Abstract: Each of DAW processes and UI devices which are controllers of the DAW processes regularly sends a heartbeat signal including identification information of a music system which the DAW process or UI device belongs to, to a network by broadcast or multicast. The heart beat signal can be transmitted to the network using an Ethernet frame for UDP including the heartbeat signal in the data body, as a heartbeat packet. Based on the received heartbeat signal, it can be distinguished that the DAW process or UI device which sent the heartbeat signal is which of a device or music software process belonging to the relevant music system, a device or music software process belonging to another music system, and a device or music software process not belonging to any music system.

Abstract: An audio processing apparatus is configured to process audio signals from a plurality of sources. The audio processing apparatus may include a digital interface to receive status data indicating a status of at least one source, and an optical output device having a plurality of groups of graphics display areas which are respectively assigned to one of a plurality of audio channels of the audio processing apparatus. The audio processing apparatus may also include a control device configured to receive the status data, to determine at least one group of graphics display areas based on the received status data, and to control a graphics display area of the determined at least one group to display graphics generated based on the received status data.

Abstract: A system of linear equations is used to upmix a number N of audio signals to generate a larger number M of audio signals that are psychoacoustically decorrelated with respect to one another and that can be used to improve the representation of a diffuse sound field. The linear equations are defined by a matrix that specifies a set of vectors in an M dimensional space that are substantially orthogonal to each other. Methods for deriving the system of linear equations are disclosed.

Abstract: A method and speaker system that provides a single, user adjustable tool for controlling how and which spatialization parameters are used in a single speaker multi-driver system to enhance the sound stage. The tool creates the desired amount of audio surround effect to enhance the sound stage experience or effects in multiple sound recording mode, such as music or movies. The tool is represented in a scale that runs from ‘?10’ (less immersive) to ‘+10’ (more immersive) is set to taste by the listener and will vary with room acoustics and placement.

Abstract: An apparatus includes multiple microphones and a controller. The controller is coupled to receive a signal from each of the multiple microphones. The controller is configured to control a transition of the multiple microphones from an active mode to a dormant mode. When the multiple microphones are in the active mode, the controller is configured to perform signal processing responsive to signals received from at least two of the multiple microphones. When the multiple microphones are in the dormant mode, the controller is configured to select a microphone of the multiple microphones and to perform signal processing corresponding to the selected microphone while suspending signal processing corresponding to unselected microphones.

Abstract: A machine performs an ownership audit on source media files whose media tracks are used, or proposed for use, in a mash-up media file. The machine may access a first group of media identifiers from a user's media library and identify authorized media files that the user is authorized to play or use in a mash-up. The machine may access a second group of media identifiers that identify source media files selected by the user for inclusion in a mash-up. The machine may determine whether all source media files are authorized and then present a notification that indicates whether the mash-up media file can be generated, or has been generated, exclusively from authorized media files. If a source media file is not authorized, the notification may include a suggestion that the user purchase an authorization to play the media file, use it in a mash-up, or both.

Abstract: An aircraft audio panel includes a plurality of audio inputs each configured for receiving audio communications from an audio source; a plurality of audio outputs each configured for outputting the audio communications to audio equipment in the aircraft; selecting circuitry for selecting the audio communications between the audio inputs and the audio outputs; a single selector for permitting selection of one of the audio inputs or audio outputs for adjustment; and a single audio controller for permitting adjustment of an audio characteristic of the selected audio input or audio output. The audio panel may also include an audio characteristic indicator for indicating an audio characteristic of the selected audio input or audio output. The selecting circuitry may also implement a routing mode in which a user may first initiate the routing mode, then select one of the audio inputs for routing, and then select any one of the audio outputs for receiving audio communications from the selected audio input.

Abstract: A transmitter for remote control includes a first analog-to-digital converter (ADC) to receive a first audio signal from a electronic device and convert the first audio signal to a first direct-current (DC) signal, a first boost circuit connected to the first ADC to receive and amplify the first DC signal, a second ADC receives a second audio signal from the electronic device and converts the second audio signal to a second DC signal, a second boost circuit connected to the second ADC to receive and amplify the second DC signal, an energy storage element and a transmission module is powered by the energy storage element and generates a carrier signal, the transmission module receives the amplified first DC signal from the first boost circuit, the amplified first DC signal modulates the carrier signal generated by the transmission module, and the amplified second DC signal charges the energy storage element.

Abstract: An electronic music instrument comprises at least one rotary encoder with a knob configured for setting a value of a parameter of the instrument; at least one input configured for receiving at least one of a base value of the parameter and a modulation offset of the parameter; and a visual indicator associated with the encoder configured for indicating the base value of the parameter and the modulation offset of the parameter.

Abstract: The technology in one embodiment includes an audio-processing unit, for processing an input audio signal using a pre-established audio feature. The unit includes a processor, an audio-device input in operative communication with the processor, and a system-user interface being in operative communication with the processor. The unit also includes a computer-readable medium being in operative communication with the processor and comprising a tone-manipulation component and computer-executable instructions that, when executed by the processor, cause the processor to perform various operations.

Abstract: An apparatus for detecting endpoints of sound signals when sound sources vocalized from a remote site are processed even if a plurality of speakers exists and an interference sound being input from a direction different from a direction of one speaker, and a method thereof, wherein in an environment in which a plurality of sound sources exists, the existence and the length of the sound source being input according to each direction is determined and the endpoint is found, thereby improving the performance of the post-processing, and speech being input from a direction other than a direction of speech from a speaker vocalized at a remote area from a sound source collecting unit is distinguished while the speech from the speaker is being recorded, thereby enabling a remote sound source recognition without restriction on the installation region of a microphone.

Abstract: A system and method for switching between multiple audio input signals to select one of the audio input signals as audio output signal are disclosed which include receiving the audio input signals, receiving a control signal, and switching controlled by the control signal from a first one of the audio input signals to a second one of the audio input signals. Prior to switching, the first one of the audio input signals is muted and upon switching, the second one of the audio input signals is unmuted in one of multiple unmute modes based on one or more signal parameters characterizing the second one of the audio input signals.

Abstract: A frequency characteristics control device of a mixer mixes a first audio signal and a second audio signal inputted to the mixer. In the frequency characteristics control device, a characteristics detection section detects a first frequency characteristic of the first audio signal and a second frequency characteristic of the second audio signal. Based on the first and second frequency characteristics, a removal band detection section detects a removal band in which a level of the first audio signal is higher than a level of the second audio signal. A filtering process section performs a filtering process on the second audio signal so as to attenuate a component of the second audio signal in the removal band. A output section mixes the first audio signal inputted to the mixer and the second audio signal on which the filtering process section has performed the filtering process.

Abstract: One embodiment of the invention is directed to a switchable plug comprising a first incoming line configured to carry a signal from a source device, a second incoming line configured to carry a signal from the source device, a Mic Bias output line, a Ground output line, and a switch control. The switch control is operably connected with the first incoming line, the second incoming line, the Mic Bias output line and the Ground output line. The switch control is configured to detect a type of the signal carried on the first incoming line and a type of the signal carried on the second incoming line, from the source device. The switch control is further configured to automatically route a Mic Bias type signal to the Mic Bias output line and a Ground type signal to the Ground output line, without user intervention.

Abstract: An approach is provided for generating and sharing compositions based on media elements acquired from different locations. A media generation platform determines an association of one or more media elements with one or more locations. The media generation platform next determines one or more interactions with at least one connecting user interface element to associate the one or more media composition elements with one or more media composition applications. The media generation platform then causes a generation of at least one media composition that includes, at least in part, the one or more media composition elements based, at least in part, on the one or more interactions.

Abstract: Systems and methods for enhancing speech intelligibility using a companion microphone system can include microphones, a position sensor and a microcontroller. In certain embodiments, the position sensor is configured to generate position data corresponding to a position of the companion microphone system. In various embodiments, the microphones and the position sensor include a fixed relationship in three-dimensional space. In certain embodiments, the microcontroller is configured to receive the position data from the position sensor and select one or more of the microphones to receive an audio input based on the received position data.

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 being assigned with a channel and being provided with controls for adjusting values of the parameters of the assigned channel, and has 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 a setting stored in a storing section. A clearing section clears a setting stored in a storing section.

Abstract: A mixing console including a plurality of audio inputs and a plurality of audio processing channels. Control data is received from a microphone which is connected to a given audio input to provide audio data. The control data includes an indication of an audio source associated with a microphone. A router of the mixing console is configured to route the audio data from the given audio input to a given audio processing channel based on the received indication of the audio source.

Abstract: A system and method for enhancing comprehensibility through spatialization may receive two or more audio signals each associated with one of two or more audio sources. A respective panning gain may be calculated for each of the two or more audio signals using a count of the total number of audio sources and a count of a total number of output channels. The respective panning gain may be calculated further responsive to audio source metadata associated with each of two or more audio sources. Each of the two or more audio signals may be gain adjusted responsive to the respective panning gain. Each of the two or more gain adjusted audio signals may be mixed to create two or more output channels wherein a reproduction of the output channels produces enhanced comprehensibility.

Abstract: An effective and simple psychoacoustic bass generation system generates a harmonic signal having inter-modulation controllable to remain below a threshold level and includes a high-pass filter configured to pass harmonics which are reproducible with fidelity by the loudspeaker or other transducer and a loudness matching block configured to compensate the loudness of the desired harmonics to match the loudness of the original signal.