Abstract: An apparatus for audio signal processing audio objects within at least one audio scene, the apparatus comprising at least one processor configured to: define for at least one time period at least one contextual grouping comprising at least two of a plurality of audio objects and at least one further audio object of the plurality of audio objects outside of the at least one contextual grouping, the plurality of audio objects within at least one audio scene; and define with respect to the at least one contextual grouping at least one first parameter and/or parameter rule type which is configured to be applied with respect to a common element associated with the at least two of the plurality of audio objects and wherein the at least one first parameter and/or parameter rule type is configured to be applied with respect to individual element associated with the at least one further audio object outside of the at least one contextual grouping, the at least one first parameter and/or parameter rule type being appli

Abstract: Distributed audio mixing may include transmitting a set of parameters from a local location to one or more remote locations at least multiple miles away from the local location for, at each of the one or more remote locations, one or more remote audio sources to be processed according to the parameters to produce respective one or more remote audio mixes; processing one or more local audio sources according to the parameters to produce a local audio mix; receiving the one or more remote audio mixes; and locally summing the one or more remote audio mixes to the local audio mix to obtain a final audio mix.

Abstract: A method of generating a virtual reality (“VR”) experience includes detecting, using external sensors, a real-world spectator that is in a spectator space that is adjacent to a user that is wearing a VR headset and experiencing a VR space. The method also includes detecting a real-world physical state of the spectator, and rendering, in response to detecting, the spectator in the VR space as an avatar that reflects the real-world physical state of the spectator.

Abstract: Methods and systems are provided for independent audio volume control. User input may be received in an audio setup configured for combined-audio, with the user input including one or both of a selection of a desired volume of a first type audio component of the combined-audio signal, and a selection of a desired volume of a second type audio component of the combined-audio signal. Based on the user input, a corresponding mix setting may be determined, with the mix setting including one or more control parameters for adaptively controlling mixing of the first type audio component and the second type audio component to generate the combined-audio signal. The mix setting may be determined based on a data set characterizing an audio device where at least portion of the mixing is performed. The data set may be pre-programmed and/or determined based on testing of the audio device.

Abstract: A display method includes displaying a list of a plurality of groups in a first area which is a first partial area of a display, and a name of a component belonging to at least one of the plurality of groups in a second area which is a second partial area of the display and is different from the first area. In the displaying, switching a display mode between a first display mode and a second display mode is performed in the second area. The first display mode is a mode in which all of the plurality of groups are displayed and a group, out of the plurality of groups, to which the component belongs is displayed in a display appearance different from that of a different group of the plurality of groups. The second display mode is a mode in which, out of the plurality of groups, only the group to which the component belongs is displayed.

Abstract: An apparatus for estimating an inter-channel time difference between a first channel signal and a second channel signal, includes: a calculator for calculating a cross-correlation spectrum for a time block from the first channel signal in the time block and the second channel signal in the time block; a spectral characteristic estimator for estimating a characteristic of a spectrum of the first channel signal or the second channel signal for the time block; a smoothing filter for smoothing the cross-correlation spectrum over time using the spectral characteristic to obtain a smoothed cross-correlation spectrum; and a processor for processing the smoothed cross-correlation spectrum to obtain the inter-channel time difference.

Abstract: A method of managing at least one audio playback equipment group including at least one piece of audio playback equipment is provided with a light-emitting identification interface. The method includes the steps of: using the light-emitting identification interface of the piece of audio playback equipment to display a light pattern characteristic at least of the group to which said piece of equipment belongs and/or characteristic of the role of the piece of equipment within said group in the event of at least one of the pieces of audio playback equipment being selected, and performing an action relating to the associated piece of audio playback equipment.

Abstract: An operation panel for an audio processing apparatus includes a plurality of components including a user interface that selects at least one output channel among a plurality of output channels and presents a first processing parameter, which indicates a send level from an input channel to the selected at least one output channel, and a second processing parameter different from the first processing parameter. The audio processing apparatus further includes a link receiver that receives, for at least two components, among the plurality of components, for setting a first link mode and a second link mode independently the first link mode for linking the presented first processing parameter, and the second link mode for linking the presented second processing parameter.

Abstract: The present invention relates generally to the field of digital content, and more particularly to an improved system and methods for producing, mixing, and recording content. In particular, the system may include a plurality of input channels to which one or more audio sources may be linked via a user interface. Each audio source may be assigned to a physical fader or a virtual fader for adjusting a level of an audio signal received from the audio source. Advantageously, the system may connect via a wired and/or wireless connection with a number of audio devices and may further facilitate assigning each audio source to an input channel such that content creators, such as podcasters, are able to produce, mix, and record audio contents more efficiently and effectively.

Abstract: A method for configuring a modular safety switching apparatus having electronic modules comprises: a) selecting a configuration data set of the apparatus from a database and transmitting the configuration data set to a computing apparatus, wherein the configuration data set is processed by a computer program to generate a target configuration of the apparatus and the electronic modules; b) displaying the target configuration; c) plugging one of the electronic modules into one of the module slots of the module row; d) setting a rotary switch position of one or more rotary switches of the electronic module; e) determining an actual configuration of the electronic module configured according to c) and d); f) comparing the actual configuration with the target configuration; g) displaying whether a deviation exists between the actual configuration and the target configuration; and h) repeating c) to g) for each additional electronic module.

Abstract: The present invention provides a signal processing device that allows a user, who adjusts a signal level to match a reference level, to adjust the signal level while checking a degree of difference between the signal level and the reference level. A signal processing device according to the present invention comprises: an input part 20 to which an input signal is input; a storage 40 that stores therein one or more setting values of a parameter to be used for amplifying the input signal and a reference value of the setting value; an amplification part 30 that generates an output signal by amplifying a level of the input signal based on the setting value; an output part 70 that outputs the output signal; an adjustment part 50 that adjusts the setting value; and a display 60 that displays a display result determined based on the setting value adjusted by the adjustment part and the reference value.

Abstract: An output buffer set is coupled to a first pair of pins and configured to output art audio output signal to the first pair of pins. A direct current (DC) shifting buffer set is coupled between the audio codec circuit and the first pair of pins, and is directly coupled to the output buffer set and the first pair of pins. An audio port is coupled to the first pair of pins. The DC shifting buffer set is configured to receive a first audio input signal from the audio port through the first pair of pins. The audio codec circuit is configured to record the audio output signal back and mix the recorded audio output signal with a second audio input signal, or mix the first audio input signal with a third audio input signal.

Abstract: A real-time service playback method includes, during running a real-time service, in response to receiving an input instruction, obtaining a playback speed and a time coordinate of a first anchor point, the time coordinate of the first anchor point being a start point of playback, and in response to the input instruction, stopping playing a real-time content of the real-time service, based on the time coordinate of the first anchor point, calculating a time coordinate of a second anchor point in future, and reading and storing a real-time content between a time coordinate value of the first anchor point and a time coordinate value of the second anchor point to play according to the playback speed until the playback reaches the time coordinate value of the second anchor point to resume the real-time service. The time coordinate of the second anchor point is an end point of the playback.

Abstract: In at least one embodiment, an active noise cancellation system. The system includes a first controller, a data bus, and a second controller. The first controller receives first information from a plurality of noise sensing devices and second information from an audio system. The first controller employs a time division multiplexing scheme to generate a multiplexed stream of data including the first information and the second information. The data bus transmits the multiplexed stream of data on a single data channel. The second controller receives the multiplexed stream of data on the single data channel and separates the first information from the second information on the multiplexed stream of data to perform ANC functionality.

Abstract: Mixer devices for efficiently processing multitrack audio including: at least one audio input port for receiving local audio from an audio input device as at least one local audio input; a memory; a user interface, the user interface including at least one adjuster; and at least one control unit electrically coupled to said receiver, said transmitter; said at least one audio input port, said memory, and said user interface; the at least one control unit programmed to control local gain of the local audio and local mixing. In another aspect, the mixer devices also remotely control the gain of bodypack receivers capable of receiving local audio input in analog or digital form. A novel display is also incorporated to facilitate such control surrounding the adjustor capable of performing the control.

Abstract: An audio processing device includes: a sound level measuring circuit arranged to operably generate multiple sound level values, wherein the multiple sound level values respectively correspond to the sound levels generated by an audio playback device at multiple time points or the sound levels received by a microphone at multiple time points; an audio dose calculating circuit coupled with the sound level measuring circuit and arranged to operably generate an audio dose value corresponding to a measuring period based on the multiple sound level values and contents of a weighting table; a control circuit coupled with the audio dose calculating circuit and arranged to operably compare the audio dose value with a dose threshold to determine whether to generate a control signal or not; and an indication signal generating circuit coupled with the control circuit and arranged to operably generate a corresponding indication signal according to the control signal.

Abstract: Methods, systems, and computer program products for network-based processing and distribution of multimedia content of a live performance are disclosed. In some implementations, recording devices can be configured to record a multimedia event (e.g., a musical performance). The recording devices can provide the recordings to a server while the event is ongoing. The server automatically synchronizes, mixes and masters the recordings. The server performs the automatic mixing and mastering using reference audio data previously captured during a rehearsal. The server streams the mastered recording to multiple end users through the Internet or other public or private network. The streaming can be live streaming.

Abstract: The present technology relates to a signal processing apparatus and method, and a program that can easily determine a localization position of a sound image. A signal processing apparatus includes: an acquisition unit configured to acquire information associated with a localization position of a sound image of an audio object in a listening space specified in a state where the listening space viewed from a listening position is displayed; and a generation unit configured to generate a bit stream on the basis of the information associated with the localization position. The present technology can be applied to the signal processing apparatus.

Abstract: A system comprises an auditory device processor, an auditory device output mechanism, an auditory input sensor, a database including a reference bank of environmental sounds and corresponding sound profiles, and a memory.

Abstract: The present invention provides for an analog audio patchbay comprising a first plurality of connectors capable of receiving a plurality analog electronic signals, a second plurality of connectors capable of transmitting a plurality of analog electronic signals, and a circuit board extending from the first plurality to the second plurality.

Abstract: Sound device receives a first sound signal from a first sound processor, generates a second sound signal based on the first sound signal, and transmits the second sound signal to a second sound processor. The second sound processor performs signal processing to the second sound signal to generate a third sound signal. The sound device receives the third sound signal from the second sound processor, checks a state of the second sound processor based on a signal received from the second sound processor, transmits a fourth sound signal based on the third sound signal to the first sound processor when determining that the state of the second sound processor is normal, and generates a fifth sound signal based on the first sound signal or the second sound signal to transmit it to the first sound processor when determining that the state of the second sound processor is abnormal.

Abstract: A system for generating channel-compensated features of a speech signal includes a channel noise simulator that degrades the speech signal, a feed forward convolutional neural network (CNN) that generates channel-compensated features of the degraded speech signal, and a loss function that computes a difference between the channel-compensated features and handcrafted features for the same raw speech signal. Each loss result may be used to update connection weights of the CNN until a predetermined threshold loss is satisfied, and the CNN may be used as a front-end for a deep neural network (DNN) for speaker recognition/verification. The DNN may include convolutional layers, a bottleneck features layer, multiple fully-connected layers and an output layer. The bottleneck features may be used to update connection weights of the convolutional layers, and dropout may be applied to the convolutional layers.

Abstract: A system for generating audio content in dependence upon an input audio track comprising audio corresponding to one or more sound sources, the system comprising an audio input unit operable to input the input audio track to one or more models, each representing one or more of the sound sources, and an audio generation unit operable to generate, using the one or more models, one or more audio tracks each comprising a representation of the audio contribution of the corresponding sound sources of the input audio track, wherein the generated audio tracks comprise one or more variations relative to the corresponding portion of the input audio track.

Abstract: A hearing device includes: first and second primary microphones for provision of first and second primary microphone input signals; a secondary microphone for provision of a secondary microphone input signal; a mixing module for provision of a mixer output based on a primary mixer input and/or a secondary mixer input, primary mixer input being based on the first and second primary microphone input signals, the secondary mixer input being based on the secondary microphone input signal; a processing unit configured to provide an electrical output signal; and a receiver for converting the electrical output signal to an audio output signal; wherein the mixing controller is configured to increase an amount of the secondary mixer input in the mixer output for a time period after a presence of a sound event is determined, and reduce the amount of the secondary mixer input in the mixer output after the time period.

Abstract: An audio signal processing method performs signal processing on a first audio signal to be outputted to a first device that a performer uses, the first audio signal on which the signal processing has been performed being a second audio signal, receiving a setting that causes the first audio signal to send to a monitor bus which is for to output the second audio signal, and performing signal processing on the second audio signal, which is received via the monitor bus and is to be outputted to a second device different from the first device, such that a sound quality of a sound to be outputted by the second device is closer to sound quality of a sound to be outputted by the first device than in a case where the signal processing is not performed on the second audio signal.

Abstract: Systems and methods for programmable pulse density modulation (PDM) components enable backwards compatibility while maintaining reasonable tolerances. A system includes a programmable PDM device, a PDM master device and a bus communicably coupling the programmable PDM device to the PDM receiver. The PDM device may include an audio sensor, audio input circuitry, a delta-sigma converter and a PDM transmitter and receiver. The PDM transmitter and receiver may send out PDM data from the PDM device and receive programming data from the PDM Master device. The PDM device may further include register space controlled by the PDM master device, a buffer storing audio data for wakeup word systems that store audio data when the PDM receiver is powered down, a bus holder to hold the previous value on the bus if no device is driving it, and/or a clock multiplier to multiply the incoming clock by a factor.

Abstract: A digital mixer is disclosed having a plurality of displays and which applies a mixing process on an audio signal. The digital mixer includes an input interface, one or more processors, an output interface, at least a first display and a second display, and an operator. The operator includes a mode switching button for switching between a first mode and a second mode. The one or more processors are configured to cause parameter setting screens of audio signals of different channels to be displayed respectively on the first display and the second display when the mode switching button is set to the first mode, and cause different parameter setting screens of an audio signal of a single channel to be displayed respectively on the first display and the second display when the mode switching button is set to the second mode.

Abstract: A microphone system, comprises a first transducer, for generating a first acoustic signal, and a second transducer, for generating a second acoustic signal. A high-pass filter receives the first signal and generates a first filtered signal, and a low-pass filter receives the second signal and generates a second filtered signal. An adder forms an output signal of the microphone system as a sum of the first filtered signal and the second filtered signal.

Abstract: A signal processor and a signal processing method thereof. The signal processor obtain a first audio signal of a first channel, obtains a second audio signal of a second channel. The signal processor performs a first signal processing on the input first audio signal. The signal processor, when it does not obtain the second audio signal, performs a second signal processing on the input first audio signal having undergone the first signal processing and output a further processed first audio signal.

Abstract: An audio signal processing apparatus includes a first bus, a second bus, and a controller that assigns a first input channel to the first bus, assigns a second input channel to the second bus, receives an amount of feed from the first input channel to the first bus, and receives an amount of feed from the second input channel to the second bus.

Abstract: The present disclosure relates to multimedia data playing methods and terminal devices. One example method includes obtaining, by a terminal device, a first sound effect plug-in, where the first sound effect plug-in corresponds to a first sound effect, and the terminal device includes a first application and a second application that are used to play multimedia data, receiving, by the terminal device, a first instruction used to indicate that a user of the terminal device selects the first sound effect option, wherein where the first sound effect option is an option corresponding to the first sound effect, when first multimedia data is to be played by the first application or the second application, processing, by the terminal device, the first multimedia data by using the first sound effect plug-in according to the first instruction to play the first multimedia data with the first sound effect.

Abstract: A method for playing audio source, executed by one or more processors, comprising: playing an audio source set including a first audio source and a second audio source; displaying a first object for an interaction with a user; obtaining an input for the first object; and adjusting at least a part of the audio source set based on the input; wherein in order that a selective change to the audio source set is given by the interaction with the user, when a first input related to the first object is obtained, the audio source set is adjusted so that an output level of the first audio source selected by the first input is highest, wherein in order that a random change to the audio source set is given by the interaction with the user, when a second input related to the first object is obtained, a random audio effect is applied to at least one audio source included in the audio source set.

Abstract: A system reduces sound coloration caused by rendering of a 3D audio signal. The system renders the 3D audio signal including a plurality of channels using the input audio signal. Input spectra data defining spectral information of the input audio signal is computed. 3D spectra data defining spectral information of a single channel representation of the 3D audio signal is computed. The system generates a tonal balance filter based on the input spectral data and the 3D spectral data. The tonal balance filter, when applied to the 3D audio signal, reduces sound coloration caused by the rendering of the 3D audio signal. The tonal balance filter is applied to the 3D audio signal to generate an output audio signal and the output audio signal is presented via a speaker array.

Abstract: Techniques for concurrent transmission of audio and ultrasound are described. In an example, a computing device generates, in a digital domain, mixed audio data from multiple sets of audio data, each set corresponding to a different audio channel. The computing device also generates, in the digital domain, ultrasound data, and generates serial data by providing the mixed audio data and the ultrasound data as different inputs to an I2S mixing module. In an analog domain, the computing device generates an output signal based at least in part on the serial data, and sends the output signal to a speaker.

Abstract: Some methods involve receiving an input audio signal that includes N input audio channels, the input audio signal representing a first soundfield format having a first soundfield format resolution, N being an integer ?2. A first decorrelation process may be applied to two or more of the input audio channels to produce a first set of decorrelated channels, the first decorrelation process maintaining an inter-channel correlation of the set of input audio channels. A first modulation process may be applied to the first set of decorrelated channels to produce a first set of decorrelated and modulated output channels. The first set of decorrelated and modulated output channels may be combined with two or more undecorrelated output channels to produce an output audio signal that includes O output audio channels representing a second and relatively higher-resolution soundfield format than the first soundfield format, O being an integer ?3.

Abstract: A parameter setting apparatus includes a first display device, a first operation controller, a mode operation controller configured to receive either designation of a first adjustment mode for adjusting a send level or a second adjustment mode for adjusting a first parameter other than the send level, and a processing controller. The controller includes a processor. The processor controls the first display device to display a bus selection screen on the display device, selects a bus based on an operation on the displayed selection screen, causes the first operation controller to adjust a send level of the selected bus upon the mode operation controller receiving the designation of the first adjustment mode, and causes the first operation controller to adjust the first parameter upon the mode operation controller receiving the designation of the second adjustment mode.

Abstract: A system, method and computer product for combining audio tracks. In one example embodiment herein, the method comprises determining at least one music track that is musically compatible with a base music track, aligning those tracks in time, and combining the tracks. In one example embodiment herein, the tracks may be music tracks of different songs, the base music track can be an instrumental accompaniment track, and the at least one music track can be a vocal track. Also in one example embodiment herein, the determining is based on musical characteristics associated with at least one of the tracks, such as an acoustic feature vector distance between tracks, a likelihood of at least one track including a vocal component, a tempo, or musical key. Also, determining of musical compatibility can include determining at least one of a vertical musical compatibility or a horizontal musical compatibility among tracks.

Abstract: Systems and methods for generating a playback-information display during time compression or expansion of an audio signal are provided. The system includes a processor that performs a method including displaying a first remaining playback-time associated with an audio file; adjusting the playback speed of the audio file during playback of the audio file; and, in response to the playback speed being adjusted, automatically displaying a second remaining playback-time associated with the audio file during playback of the audio file.

Abstract: Some embodiments of the invention are directed to enabling a user to easily identify the frequency range(s) at which sound masking occurs, and addressing the masking, if desired. In this respect, the extent to which a first stem is masked by one or more second stems in a frequency range may depend not only on the absolute value of the energy of the second stem(s) in the frequency range, but also on the relative energy of the first stem with respect to the second stem(s) in the frequency range. Accordingly, some embodiments are directed to modeling sound masking as a function of the energy of the stem being masked and of the relative energy of the masked stem with respect to the masking stem(s) in the frequency range, such as by modeling sound masking as loudness loss, a value indicative of the reduction in loudness of a stem of interest caused by the presence of one or more other stems in a frequency range.

Abstract: A method for live manipulation of signal flows via a controller. The method includes feeding in a first signal flow and a further signal flow, each having X signal flow layers, where X is greater than 2. The method further includes separating the signal flow layers from each signal flow into a respective series of sub-signal flows, related to the signal flow, as according to a predetermined ratio, where each sub-signal flow has Y sub-signal flow layers, and where Y is smaller than X. The method includes reading a desired ratio between the first signal flow and the further signal flow via a controller. The method includes merging corresponding sub-signal flows as according to the desired ratio in order to obtain a modified series of sub-signal flows. The method includes feeding out the modified series.

Abstract: The present invention provides for an analog audio patchbay comprising a first plurality of connectors capable of receiving a plurality analog electronic signals, a second plurality of connectors capable of transmitting a plurality of analog electronic signals, and a circuit board extending from the first plurality to the second plurality.

Abstract: In general, techniques of controlling a plurality of multimedia players in groups are disclosed. According to one aspect of the present invention, a mechanism is provided to allow a user to group some of the players according to a theme or scene, where each of the players is located in a zone. When the scene is activated, the players in the scene react in a synchronized manner. For example, the players in the scene are all caused to play a multimedia source or music in a playlist, wherein the multimedia source may be located anywhere on a network.

Abstract: Methods, systems, and computer program products of automatic de-essing are disclosed. An automatic de-esser can be used without manually setting parameters and can perform reliable sibilance detection and reduction regardless of absolute signal level, singer gender and other extraneous factors. An audio processing device divides input audio signals into buffers each containing a number of samples, the buffers overlapping one another. The audio processing device transforms each buffer from the time domain into the frequency domain and implements de-essing as a multi-band compressor that only acts on a designated sibilance band. The audio processing device determines an amount of attenuation in the sibilance band based on comparison of energy level in sibilance band of a buffer to broadband energy level in a previous buffer. The amount of attenuation is also determined based on a zero-crossing rate, as well as a slope and onset of a compression curve.

Abstract: In general, techniques of controlling a plurality of multimedia players in groups are disclosed. According to one aspect of the present invention, a mechanism is provided to allow a user to group some of the players according to a theme or scene, where each of the players is located in a zone. When the scene is activated, the players in the scene react in a synchronized manner. For example, the players in the scene are all caused to play a multimedia source or music in a playlist, wherein the multimedia source may be located anywhere on a network.

Abstract: A computer system is trained to understand audio-visual spatial correspondence using audio-visual clips having multi-channel audio. The computer system includes an audio subnetwork, video subnetwork, and pretext subnetwork. The audio subnetwork receives the two channels of audio from the audio-visual clips, and the video subnetwork receives the video frames from the audio-visual clips. In a subset of the audio-visual clips the audio-visual spatial relationship is misaligned, causing the audio-visual spatial cues for the audio and video to be incorrect. The audio subnetwork outputs an audio feature vector for each audio-visual clip, and the video subnetwork outputs a video feature vector for each audio-visual clip. The audio and video feature vectors for each audio-visual clip are merged and provided to the pretext subnetwork, which is configured to classify the merged vector as either having a misaligned audio-visual spatial relationship or not.

Abstract: An audio signal processor includes a first block including a first input interface and a first output interface, and a second block including a second input interface and a second output interface. The audio signal processor further includes a routing matrix that is connected to each of the first input interface, the first output interface, the second input interface, and the second output interface. The routing matrix outputs a signal inputted from each of the first output interface and the second output interface, to each of the first input interface and the second input interface.

Abstract: A mixing apparatus having a stereo output includes: a first signal processor that mixes a first signal and a second signal in a first channel; a second signal processor that mixes a third signal and a fourth signal in a second channel; a third channel that processes a weighted sum of a signal of the first channel and a signal of the second channel; and a gain deriving part that generates a gain mask commonly used in the first channel and the second channel, wherein the gain deriving part determines a first gain commonly applied to the first signal and the third signal, and a second gain commonly applied to the second signal and the fourth signal, so that predetermined conditions for simultaneous gain generation are satisfied at least at the first channel and the second channel among the first channel, the second channel, and the third channel.

Abstract: The present invention relates to a method for processing and playing audio data comprising the steps of receiving mixed input data and playing recombined output data. Furthermore, the invention relates to a device for processing and playing audio data, preferably DJ equipment, comprising an audio input unit for receiving a mixed input signal, a recombination unit and a playing unit for playing recombined output data. In addition, the present invention relates to a method and a device for representing audio data, i.e. on a display.

Abstract: A signal processor includes a memory storing instructions and a processor that implements the stored instructions to execute a plurality of tasks, the tasks including a first input task configured to obtain a first audio signal of a first channel, a second input task configured to obtain a second audio signal of a second channel, a first signal processing task configured to perform a first signal processing on the input first audio signal, a second signal processing task configured to perform a second signal processing on the input second audio signal, and a control task configured to, when the second input task does not obtain the second audio signal, cause the second signal processing task to perform the second signal processing on the input first audio signal having undergone the first signal processing by the first signal processing task.

Abstract: A system and method for music delivery and customization includes a concept and approach to music recording, storage, customization, and delivery, in a manner such that subsequent listeners can individually supplement and/or adjust music content for later playback or other uses. Music can be arranged by merging any number of recorded tracks with live music and rerecorded or remixed, where control can be achieved in real time and in the further presence of video.