Abstract: An electronic device having a display section, comprising: a processor, wherein the processor executes the processing of: displaying a module image corresponding to a module used for sound processing and a sound information image corresponding to sound information relating to generation of a sound which can be set in the module on the display section; receiving an operation on at least one of the sound information image displayed on the display section and the module image displayed on the display section; and controlling the sound processing performed by setting the sound information corresponding to the sound information image to the module according to the operation.

Abstract: A system and method for differentially locating and modifying audio sources that includes receiving multiple audio inputs from a set of distinct locations; determining a multi-dimensional audio map from the audio inputs; acquiring a set of positional audio control inputs applied to the audio map, each audio control input comprising a location and audio processing property; and generating an audio output according to the audio control inputs and the audio inputs. The audio control inputs capable of configuration through manual, automatic, computer vision analysis, and other configuration modes.

Abstract: An apparatus comprising: an input configured to receive at least one audio signal input from at least one microphone; at least one microphone configuration determiner configured to provide for the at least one microphone a location on the apparatus; at least one sensor configured to provide at least one orientation of the apparatus; a recording mode determiner configured to determine at least one recording mode for the apparatus based on the location of the at least one microphone and the at least one orientation of the apparatus; a recording mode controller configured to determine at least one recording parameter for the at least one audio signal input from the at least one microphone based on the at feast one recording mode; and a digital signal processor configured to apply the at least one recording parameter to the at least one audio signal input.

Abstract: A library stores a plurality of presets, each of the preset being a parameter set including a plurality of parameters defining operation of a channel. When an operation to open the library is performed in a setting screen of a processing block of a selected channel, a library screen of the certain processing block is opened. A list of the plurality of presets and a block recall button are displayed on the library screen. When a user selects one preset from the list and operates the block recall button, only a parameter sub-set of the certain processing block is recalled from among the selected preset.

Abstract: Examples described herein involve determining an audio processing algorithm for a playback device to apply when playing audio content in a playback zone. Determining the audio processing algorithm may involve causing the playback device to play a first audio signal in the playback zone, receiving data indicating a second audio signal detected by a microphone of the playback device. Based on the second audio signal and a characteristic of the playback device, an audio processing algorithm may be determined. The audio processing algorithm may be calculated, or identified in a database based on an acoustic characteristic of the playback zone. The acoustic characteristic of the playback zone may be determined based on the second audio signal and the characteristic of the playback device. The audio processing algorithm may then be applied by the playback device.

Abstract: The present disclosure relates to a decoding apparatus, a decoding method, and a program that can switch, as quickly as possible, a plurality of audio encoded bit streams with synchronized reproduction timing to thereby decode and output the plurality of audio encoded bit streams. An aspect of the present disclosure provides a decoding apparatus including: an acquisition unit that acquires a plurality of audio encoded bit streams; a selection unit that determines a boundary position for switching output of the plurality of audio encoded bit streams and that selectively supplies one of the plurality of acquired audio encoded bit streams to a decoding processing unit according to the boundary position; and the decoding processing unit that applies a decoding process including IMDCT processing to the one input through the selection unit, in which the decoding processing unit skips overlap-and-add in the IMDCT processing corresponding to each frame before and after the boundary position.

Abstract: Embodiments are described for a method of rendering audio for playback through headphones comprising receiving digital audio content, receiving binaural rendering metadata generated by an authoring tool processing the received digital audio content, receiving playback metadata generated by a playback device, and combining the binaural rendering metadata and playback metadata to optimize playback of the digital audio content through the headphones.

Abstract: A device for audio leveling and sound enhancement to overcome the lower sounding level than an audio video transmission. The device is interposed at source, such an HDMI cable and a device that can reproduce the enhanced sound such as a television, computer monitor and the like. If the device includes an audio leveler and a pc board or chip having the requisite circuitry and/or software for enhancing the sound and which is in electrical communication with the audio leveler. The housing includes an input port for receiving the initial audio signal and an output port which connects to a audio reproducing device after the inputted signal has been processed.

Abstract: Audio signals are processed by a real-time digital-signal-processing system. A digital-audio-interface is arranged to be connected to a packet-based-network. A control-processor arranged presents a graphical-interface to a user and an audio-source is connected to the packet-based-network. The control-processor receives an instruction to connect the external-audio-source to the digital-signal-processing system via the packet-based-network. In response to receiving the instruction, the control-processor establishes a first-connection-node between the audio-source and the packet-based-network; creates a second-connection-node between the packet-based-network and the digital-audio-interface; and configures a third-connection-node between the digital-audio-interface and the digital-signal-processing system.

Abstract: The present invention concerns a method and apparatus for the modulation of an acoustic field for providing tactile sensations. A method of creating haptic feedback using ultrasound is provided. The method comprises the steps of generating a plurality of ultrasound waves with a common focal point using a phased array of ultrasound transducers, the common focal point being a haptic feedback point, and modulating the generation of the ultrasound waves using a waveform selected to produce little or no audible sound at the haptic feedback point.

Abstract: A method and apparatus is provided that controls other electronic devices through one electronic device in a communication system. The electronic device for executing control commands includes a communication unit configured to communicate with at least one of other electronic devices, a memory configured to store identification information about other electronic devices to execute a group control command, and a controller configured to execute, when receiving the control command, the received control command, determine whether the received control command is a group control command that other electronic devices need to execute in the same way as the electronic device, read, when the received control command is a group control command, identification information about at least one of other electronic devices as group control objects from the memory, and transmit the identification information to other electronic devices.

Abstract: An audio signal processing device, including: input ports; channels; a first input patch to patch one of the input ports to one of the channels; a second input patch to patch one of the input ports to one of the input channels, wherein whether switching is enabled or disabled is set for each channel; and a mode selector to select a first mode or a second mode, wherein, in the first mode, the one input channel receives audio signals of the one input port patched in the first input patch, and wherein, in the second mode, the one channel set to be enabled receives audio signals of the one input port patched in the second input patch while the one channel set to be disabled receives audio signals of the one input port patched in the first input patch.

Abstract: A system and method for differentially locating and modifying audio sources that includes receiving multiple audio inputs from a set of distinct locations; determining a multi-dimensional audio map from the audio inputs; acquiring a set of positional audio control inputs applied to the audio map, each audio control input comprising a location and audio processing property; and generating an audio output according to the audio control inputs and the audio inputs. The audio control inputs capable of configuration through manual, automatic, computer vision analysis, and other configuration modes.

Abstract: Sound equipment includes an operation element, a memory, a processor, a controller, and a display. The operation element receives a set value. The memory stores the set value. The processor performs processing by the set value of the memory. The controller executes a normal state in which the set value of the memory is updated by the set value of the operation element, and a locked state in which the set value of the memory is fixed regardless of the set value of the operation element, and, by either a first method when the set values are equal or a second method by updating the set value of the memory with the set value of the operation element, releases the locked state and shifts the locked state to the normal state. The display displays a selection screen that receives a selection of the first method or the second method.

Abstract: A management device has a plurality of tracks (MTR) corresponding to a plurality of terminal devices, commands the terminal devices to start/stop recording in response to recording start/stop instructions (recording command circuitry), and creates, in each of the tracks, a provisional segment identifying a recording time corresponding to the instructions (provisional segment creation circuitry). Each of the terminal devices records sound data in response to the recording start/stop instructions (recording circuitry) and transfers the recorded sound data to the management device (transfer circuitry). The management device 10 acquires the sound data from the individual terminal devices (acquisition circuitry), and then the management device records the acquired sound data in association with the provisional segments of the corresponding tracks and thereby converts the provisional segments into actual segments (conversion circuitry).

Abstract: A method of selectively modifying an initial audio output of a device that has at least two speakers is provided. The method comprises detecting a volume level of the initial audio output; comparing the volume level to a threshold; based on the comparison, controlling reproduction of the initial audio output by a selective execution of: (i) responsive to the volume level being inferior to the threshold, transmitting an identical audio signal to each one of the speakers for reproducing a modified audio output being of a mono audio output type; and (ii) responsive to the volume level being superior to the threshold, transmitting respective audio signals to the speakers for reproducing the modified audio output, the respective audio signals being different from one another and where the modified audio output is of a stereo audio output type.

Abstract: A vehicle includes one or more microphones that are each configured to selectively provide sound signals from one or more seating positions in the vehicle. A controller is programmed to receive an input indicative of a request to enter a group conversation mode upon initiation of a call, and, responsive to receiving the input, enable the microphones to selectively provide sound signals from all of the seating positions.

Abstract: An audio-mixing processing method, an audio-mixing processing apparatus, a device, and a storage medium are provided. The method includes: extracting a signal sm(n) of a frame in a signal of a first channel and a signal vm(n) of a frame in a signal of a second channel, where the signal vm(n) corresponds to the signal sm(n), the signal of the second channel is required to be mixed with the signal of the first channel, and n represents the number of sample points; weighting the signal sm(n) and the signal vm (n), and adding the weighted signal sm(n) and the weighted signal vm(n) to acquire a mixed signal ym(n); and calculating a variable attenuation factor based on a maximum value ymax of a sequence of the mixed signal ym(n), and attenuating the mixed signal ym(n) by using the variable attenuation factor to generate an output signal Zm(n).

Abstract: Methods for generating an object based audio program, renderable in a personalizable manner, and including a bed of speaker channels renderable in the absence of selection of other program content (e.g., to provide a default full range audio experience). Other embodiments include steps of delivering, decoding, and/or rendering such a program. Rendering of content of the bed, or of a selected mix of other content of the program, may provide an immersive experience. The program may include multiple object channels (e.g., object channels indicative of user-selectable and user-configurable objects), the bed of speaker channels, and other speaker channels. Another aspect is an audio processing unit (e.g., encoder or decoder) configured to perform, or which includes a buffer memory which stores at least one frame (or other segment) of an object based audio program (or bitstream thereof) generated in accordance with, any embodiment of the method.

Abstract: A communication system includes a first microphone, a first loudspeaker, a second loudspeaker, a second microphone, a speaker number determination unit, and a control unit. The first microphone acquires a surrounding voice as a voice signal. The first loudspeaker is provided near the first microphone and outputs a voice. The second loudspeaker acquires the voice signal from the first microphone and outputs the voice signal as a voice. The second microphone acquires a surrounding voice as a voice signal and outputs the voice signal to the first loudspeaker. The speaker number determination unit determines the number of speakers around the first microphone based on the voice signal acquired by the first microphone. The control unit controls a volume of the first loudspeaker based on the determination of the speaker number determination unit.

Abstract: The present disclosure relates to a method of downmixing a plurality of input audio channels. The method include obtaining, for each of the input audio channels, a plurality of frequency coefficients in a plurality of corresponding frequency bins, and applying, for at least one frequency bin, a downmix matrix to a first array formed by the frequency coefficients of the plurality of input audio channels for the respective frequency bin to obtain a second array formed by the frequency coefficients of a plurality of intermediate audio channels for the respective frequency bin. The method further involves determining a third array including only the non-zero entries of the downmix matrix, and determining a fourth array including, for each entry of the third array, an entry indicative of a position of the respective entry of the third array within the downmix matrix.

Abstract: Embodiments are described for a method of rendering audio for playback through headphones comprising receiving digital audio content, receiving binaural rendering metadata generated by an authoring tool processing the received digital audio content, receiving playback metadata generated by a playback device, and combining the binaural rendering metadata and playback metadata to optimize playback of the digital audio content through the headphones.

Abstract: A vehicle is provided having a passenger compartment with at least a front seat position in a front audio zone and at least a rear seat position in a rear audio zone. The vehicle includes at least left and right front speakers respectively in relation to the forward direction of the vehicle, which are arranged in front of the front seat position; left and right center speakers respectively, which are arranged in front of the rear seat position; a central front speaker which is arranged centrally in front of the front seat position; and a control unit which is designed to route signal portions of individual channels of a multi-channel audio signal to the speakers such that the multi-channel audio signal is spatially played back in the front and the rear audio zone based on the seat positions; wherein the front seat position is rotatable by 180 degrees.

Abstract: A device includes a processor and a transmitter. The processor is configured to determine a first mismatch value indicative of a first amount of a temporal mismatch between a first audio signal and a second audio signal. The processor is also configured to determine a second mismatch value indicative of a second amount of a temporal mismatch between the first audio signal and the second audio signal. The processor is further configured to determine an effective mismatch value based on the first mismatch value and the second mismatch value. The processor is also configured to generate at least one encoded signal having a bit allocation. The bit allocation is at least partially based on the effective mismatch value. The transmitter configured to transmit the at least one encoded signal to a second device.

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 group of one or more channels among a plurality of channels is allocated to one of channel strips. Signal-adjusting parameters of the one or more channels belonging to the group are controlled in an interlinked fashion in response to an operation of the one channel strip to which the group is allocated. In response to a deployment instruction, a deployment controlling circuitry presents values of the signal-adjusting parameters of the one or more channels belonging to the group in such a manner that the values of the signal-adjusting parameters of the one or more channels belonging to the group are presented by use of one of a display device and the channel strips, and that setting contents of manual-operators in the channel strips are presented by use of another of the display device and the channel strips.

Abstract: A device includes a processor and a transmitter. The processor is configured to determine a first value and a second value indicative of a first amount and a second amount, respectively, of a temporal mismatch between a first audio signal and a second audio signal. The processor is also configured to determine an effective value based on the first value and the second value, to select, based on the effective value, a first coding mode and a second coding mode, and to generate at least one encoded signal having a bit allocation. The at least one encoded signal is based on a first encoded signal and a second encoded signal that are based on the first coding mode and the second coding mode, respectively. The bit allocation is at least partially based on the effective mismatch value. The transmitter is configured to transmit the at least one encoded signal.

Abstract: In accordance with the present disclosure, an integrated circuit for implementing at least a portion of a personal audio device may include an output and a processing circuit. The output may provide an output signal to a transducer including both a source audio signal for playback to a listener and an anti-noise signal for countering the effect of ambient audio sounds in an acoustic output of the transducer. The processing circuit may implement an adaptive noise cancellation system that generates the anti-noise signal to reduce the presence of the ambient audio sounds heard by the listener by adapting, based on a presence of the source audio signal, a response of the adaptive noise cancellation system to minimize the ambient audio sounds at the acoustic output of the transducer, wherein the adaptive noise cancellation system is configured to adapt both in the presence and the absence of the source audio signal.

Abstract: In a method of transcoding of a video bitstream by a transcoder arrangement, performing the steps of receiving a video bitstream with a predetermined input video format, receiving side information related to the video bitstream, the side information comprising at least one of mode or motion or in-loop filter information relating to at least one other predetermined video format for the video bitstream. Further, performing the steps of decoding the received side information to generate transcoding guiding information, and encoding a representation of the received video bitstream based at least on the generated transcoding guiding information, to provide a transcoded video bitstream with a predetermined output video format.

Abstract: There is provided a method for enhancing dialog in a decoder of an audio system. The method comprises receiving a plurality of downmix signals being a downmix of a larger plurality of channels; receiving parameters for dialog enhancement being defined with respect to a subset of the plurality of channels that is downmixed into a subset of the plurality of downmix signals; upmixing the subset of downmix signals parametrically in order to reconstruct the subset of the plurality of channels with respect to which the parameters for dialog enhancement are defined; applying dialog enhancement to the subset of the plurality of channels with respect to which the parameters for dialog enhancement are defined using the parameters for dialog enhancement to provide at least one dialog enhanced signal; and subjecting the at least one dialog enhanced signal to mixing to provide dialog enhanced versions of the subset of downmix signals.

Abstract: An audio decoding system (100) for processing a two-channel input signal (X) comprises a parametric mixing stage (110). The parametric mixing stage receives the two-channel input signal and a set of mixing parameters (P1), and outputs a two-channel output signal (Y1). The parametric mixing stage comprises a decorrelation stage (111) outputting a decorrelated signal (D1) based on the input signal. The parametric mixing stage further comprises a mixing matrix (112) receiving the input signal and the de-correlated signal, and forming a two-channel linear combination of channels from the input signal and the decorrelated signal. The mixing matrix outputs the linear combination as the two-channel output signal. Coefficients of the linear combination are controllable by the set of mixing parameters, and at least four mixing parameters of the set are independently assignable.

Abstract: Audio mix information is received from a plurality of users. Mix rules are determined from the audio mix information from the plurality of users, wherein the mix rules include a first mix rule associated with a first audio item. The first mix rule relates to an overlap of the first audio item with another audio item. The first mix rule is made available to one or more clients. After making the first mix rule available, an indication, from a respective client device, that the first audio item is to be mixed with a second audio item at the respective client device in accordance with the first mix rule is received. In response to the indication, a specification of the first mix rule is transmitted to the respective client device to be applied by the respective client device to generate a transition between the first audio item and the second item.

Abstract: In an audio signal processing device including a plurality of mixing buses to mix audio signals processed in a plurality of channels, a switching instruction to switch function of the mixing buses is accepted. In accordance with the switching instruction, operation mode of one or a plurality of switch mixing buses among the plurality of mixing buses is switched between a first mode and a second mode. In first mode, the audio signal processing device controls ON/OFF of signal transmission from each of the channels to the switch mixing bus in accordance with settings of ON/OFF of signal transmission from the each channel to the switch mixing bus, and in second mode, the audio signal processing device controls ON/OFF of signal transmission from each of the channels to the switch mixing bus in accordance with manipulation of cue controls respectively provided to correspond to any of the channels.

Abstract: A visualization method for music-color synesthesia, belonging to the fields of cognitive psychology, color science and computer graphics, and relating to the visualization of the music synesthesia correlation of colors and the generation of a musical color combination scheme. In the method, a musical scale system is presented in the form of a color scale system, and the musical elements are presented through the color correlation to realize the synesthetic conversion between colors and musical elements, and on this basis, a color graphical conversion of the musical structure is performed. The implementations include: a conversion from colors to musical scales, a conversion from colors to chords, a conversion from colors to rhythm and a conversion from colors to texture etc.

Abstract: Technologies for adaptive audio communications include a telecommunications device configured to collect session data of a communication session that includes an audio stream between a user of the telecommunications device and at least one other user of a remote telecommunications device. The telecommunications device is further configured to determine a session context of the communication session based on the collected session data, determine whether the session data includes an anomaly, and adjust, in response to a determination that the anomaly was detected, at least one of a portion of the audio stream of the communication session and a setting of the telecommunications device based on the anomaly. Other embodiments are described and claimed.

Abstract: A method comprising: receiving at least one audio channel signal; receiving at least one user interface input; generating a visualization of the at least one audio channel signal dependent on the at least one user interface input; and rendering the at least one audio channel signal to generate a rendered audio signal dependent on the at least one user interface input wherein the rendered audio channel signal is substantially synchronized with the visualization of the at least one audio channel signal.

Abstract: In live music performances each musician may prefer to receive a customized signal comprising one or more signals including their own output and/or other musicians' output.

Abstract: A microphone biasing circuit comprises a first amplifier having an output connected to a first node and an input connected to a second node; and a first feedback path connected from the first node to the second node. The first feedback path comprises a microphone having a first terminal connected to the first node and a second terminal connected to a third node, the microphone being configured to provide a sensed voltage at the first node in response to sound, the third node having a first DC bias voltage; and a first capacitor connected between the third node and the second node.

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 method of controlling an audio system is disclosed, the method including: obtaining initialization configuration parameters and the identification information of each of the power amplifiers, and initializing the respective power amplifiers by the configuration parameters according to the identification information so as to obtain a corresponding frequency range of each of the power amplifiers; and when the audio data is acquired, transferring the audio data to the respective power amplifiers, so that each of the power amplifiers receives the audio data, performs digital-to-analog conversion of the audio data according to the corresponding frequency range, and further outputs the digital-to-analog converted audio data to the corresponding speaker. An apparatus for controlling an audio system is also disclosed. The present disclosure improves the sound quality of the digital television audio system and reduces the cost of the digital television audio system.

Abstract: A double buffer (four pages×2 system) constituted by a DSP program buffer (D-RAM) and a prefetch program buffer (D-RAM) is provided, and burst access is performed collectively for four samplings from the prefetch program buffer (D-RAM). When this burst access is performed, a decline in performance caused by latency that occurs each time an external memory (DDR-SDRAM) is accessed is avoided by accessing with sequential addresses.

Abstract: Systems and methods associated for providing haptic feedback are provided. For instance, data indicative of a feedback scheme associated with an input mechanism can be accessed. The feedback scheme includes one or more feedback points associated with a position of a moveable portion of the input mechanism. The input mechanism is configured to change positions responsive to an actuation of the input mechanism by a user. One or more positions of the moveable portion of the input mechanism can be determined as the input mechanism is actuated by the user. Operation of one or more feedback motors can be controlled to provide feedback to the user in accordance with the feedback scheme based at least in part on the determined one or more positions of the input mechanism.

Abstract: A system for providing digital video recorder options for editing content is disclosed. In particular, the system may include receiving requests for a first version of media content and providing access to the first version of the media content. A user that accesses the first version of the media content may select a portion of the media content, and indicate whether, for example, the portion should be skipped, modified, or emphasized when the media content is accessed on a subsequent occasion. The system, based on the selection and indication, may edit the first version of the media content to create a second version of the media content. When a subsequent attempt is made to access the first version of the media content, the system may provide the user with the option to access the second version of the media content.

Abstract: A method includes determining, using signals captured from two or more microphones (1A) configured to detect an acoustic signal from one or more sound sources, one or more prominent sound sources based on the one or more sound sources (1C-1D). The method further includes determining one or more directions relative to a position of one or more of the two or more microphones for the one or more prominent sound sources (1B-1D). The method includes modifying one or more user interface elements displayed on a user interface of a display to provide an indication at least in part of the one or more directions, relative to position of at least one microphone, of the one or more prominent sound sources (1G).

Abstract: A signal processing device includes: connecting terminals each of which is connectable to respective ones of a plurality of other signal processing devices that are different from the subject signal processing device; an analog bus connected to the connecting terminals; an input terminal connected to the analog bus and that accepts an input of an audio signal; and an output terminal connected to the analog bus and that outputs an audio signal to a sound emitting device.

Abstract: An audio playback system performs a method including identifying a first type of audio included in a first audio stream, tagging the first audio stream with a first digital tag corresponding to the first type of audio, identifying a second type of audio included in a second audio stream, tagging the second audio stream with a second digital tag corresponding to the second type of audio, rendering the first audio stream with a first equalization profile applied thereto, the first equalization profile selected responsive to the audio playback system detecting the first digital tag in the first audio stream, and rendering the second audio stream with a second equalization profile different than the first equalization profile applied thereto, the second equalization profile selected responsive to the audio playback system detecting the second digital tag in the second audio stream.

Abstract: A multitrack recording system is disclosed. A multitrack recording/reproducing apparatus records a first audio signal with a multitrack recording apparatus into any of a plurality of channels (tracks). In the meantime, a smartphone records a second audio signal as a sound record file into a recording medium. A Wi-Fi connection is established between the sound recording apparatus and the smartphone, and the sound record file is transferred from the smartphone to the recording/reproducing apparatus. The recording/reproducing apparatus stores the sound record file into a remaining channel where the first audio signal is not recorded among the plurality of channels.

Abstract: An event-drivable and programmable matrix that permits reconfigurable mixing of a first plurality of audio sources into a second audio outputs via digital control of analog-only signal paths is disclosed.

Abstract: In a recording system comprising a plurality of recording apparatuses connected in cascade to increase the number of audio channels, a desired audio signal can be monitored. In the recording apparatus, when a digital audio signal is input from a digital input terminal and an analog audio signal is input from an analog audio input terminal, the analog audio signal is converted to a digital signal, which is further embedded in the digital audio signal and output from an output terminal. When an operation unit designates a channel to be used for monitor output, among the digital audio signal input from a digital input terminal and a digital audio signal obtained by converting the analog audio signal to the digital signal, an audio signal of the channel which is designated is output for monitoring.

Abstract: Systems and methods are disclosed for detecting and removing traceable identifying audio content from electronic media content. One method includes: receiving, over a network, media content to be broadcast or distributed, the media content including audio data; determining whether the audio data of the media content includes traceable identifying audio content defined by an audio identifier; generating new audio data based on the audio data of the media content when an audio identifier is determined to be include in the audio data; removing audio data from the media content when the audio identifier is determined to be include in the audio data; adding the new audio data to the media content when the audio data is removed from the media content; and broadcasting or distributing the media content having the new audio data to one or more user devices.