Abstract: Apparatus and method to mix audio provided by a radio with audio provided by an ICS in which the audio of the radio being mixed with audio provided by the ICS and presented to one ear of a user, and in which audio provided by the ICS is presented to the other ear of the user without being mixed with audio provided by the radio.

Abstract: An apparatus comprising at least one processor and at least one memory including computer program code the at least one memory and the computer program code configured to, with the at least one processor, cause the apparatus at least to perform processing at least one control parameter dependent on at least one sensor input parameter, processing at least one audio signal dependent on the processed at least one control parameter, and outputting the processed at least one audio signal.

Abstract: An apparatus and a method for controlling, during playback, the composition of a pair of stereo separated audio signals in a stereo sound entertainment system by changing the stereo image width are disclosed. A digital mixer combines a portion of a first digital audio signal with a portion of a second digital audio signal to create a first mixed signal and a portion of the second digital audio signal with a portion of the first digital audio signal to create a second mixed signal. A listener controller independently controls, in accordance with a listener input, the portion of the first digital audio signal combined with the portion of the second digital audio signal and the portion of the second digital audio signal combined with the portion of the first digital audio signal.

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

Abstract: An array of parameter setting images corresponding to a plurality of parameters, and a plurality of tab images for selecting any one of a plurality of groups formed by grouping the plurality of parameters. The tab images are displayed on a screen in an arrangement associated with the array of the parameter setting images corresponding to the parameters belonging to corresponding ones of the groups. The grouping of the parameters is made such that a portion of the parameters included in a given one of the groups is shared with another of the groups, and a portion of the tab image corresponding to the portion of the parameters shared with the other group is displayed in a display style to indicate the sharing. Upon selecting a tab image, the parameters corresponding to the selected tab image are allocated to individual ones of the plurality of operators.

Abstract: Improved techniques for presenting sound effects at a portable media device are disclosed. The sound effects can be output as audio sounds to an internal speaker, an external speaker, or both. In addition, the audio sounds for the sound effects can be output together with other audio sounds pertaining to media assets (e.g., audio tracks being played). In one embodiment, the sound effects can serve to provide auditory feedback to a user of the portable media device. A user interface can facilitate a user's selection of sound effect usages, types or characteristics.

Abstract: A method of managing a sound source in a digital AV device and an apparatus thereof are provided. The method of managing a sound source in a digital AV device includes: extracting at least one sound source from sound being reproduced through the digital AV device; mapping an image to the extracted sound source; and managing the sound sources by using the mapped image. In addition, preferably, the extracted sound source is registered, changed, deleted, selectively reproduced, or selectively deleted by using the image. Accordingly, sound being output can be visually managed by handling the sound sources separately, a desired sound source can be selectively reproduced or removed such that utilization of the digital AV device can be enhanced.

Abstract: Two or more wireless devices can be independently controlled by their respective users, a mixer component, or a leader wireless device to perform audio recording, convert the recorded audio into a standard or proprietary audio stream format, and transmit the audio stream to a server. The real-time clocks of two or more participating wireless devices can be synchronized. A wireless device can insert timestamps into the audio stream to facilitate the mixing operation. Mixing of the two or more audio streams recorded by wireless devices can be performed by a mixer component either in real time (contemporaneously with the recording) or asynchronously with respect to the recording. The mixing can be performed in a fully automated mode, and/or in an operator-assisted mode.

Abstract: An audio mixing console includes a brightness adjusting mode selection button and an adjustment subject specification button. A user operates the brightness adjusting mode selection button to set up a brightness adjusting mode, and operates the adjustment subject specification button to specify any color LED of RGB or all colors as an adjustment subject. When an LED button is operated, brightness of an LED of a color of an adjustment subject in a multicolor LED unit incorporated in the LED button is increased one step by one step. In another example, in response to an operation of a button, brightness of an LED of a color of the adjustment subject in the multicolor LED unit incorporated in a plurality of LED buttons other than this button is increased one step by one step. This work is carried out for every LED and LED button.

Abstract: A technique for aligning a plurality of media clips is provided. One or more intra-clip points of interest (POIs) are identified in at least a first media clip. When aligning a first point in the first media clip with a second point in a second media clip, the first point may be snapped to the second point, wherein at least one of the first point and second point is an intra-clip POI. When a snap occurs, at least one of a visual or audible indication is generated, such as a “pop” sound, a snap line, or automatically aligning the first point with the second point when the first point is within a specified number of pixels of the second point. Techniques for representing multiple channels of an audio clip as a single waveform and caching waveforms are also provided.

Abstract: Techniques are described herein that are capable of spectrally shaping audio signal(s) for audio mixing. Spectrally shaping an audio signal means modifying a frequency spectrum of the audio signal. A frequency spectrum of an audio signal is a representation of the audio signal in the frequency domain. For instance, a frequency spectrum may be represented using multiple frequency bands. The frequency spectrum may be modified by modifying characteristic(s) (e.g., magnitude, phase, etc.) one or more of the frequency bands.

Abstract: An electrical signal derived from an audio source is passed through an amplifier and is then divided into two discrete signal paths. The signal traveling along the first path travels through a high pass filter and is then split so that a portion is delayed and a portion passes through a harmonics generator so that the harmonics are advanced ahead of the delayed portion from the high pass filter. The split circuit is then combined in a summing amplifier. The second portion of the signal is also delayed before it also travels to the mixer. The combined signal is passed through an amplifier and then fed into an output transducer. The use of a delay both on the input signal and the portion of the signal which passed through the high pass filter output just before the summing amplifier enables the harmonics to not be masked by the lower frequencies, making the effect more apparent.

Abstract: A connector for receiving a cylindrical plug includes a body defining a plug aperture and a cavity for receiving the cylindrical plug. A plurality of electrical contacts in communication with the cavity make electrical connections with the cylindrical plug and retain the cylindrical plug. A microphone is coupled to the body such that the plug aperture and the cavity provide an acoustic path to the microphone. The microphone may be at an end of the connector opposite the plug aperture or on a side of the connector adjacent the plug aperture. The connector allows a microphone to be added to a device, such as a mobile telephone, without the need for an additional external aperture. The microphone in the connector may operate with a second microphone and processing electronics in the device to provide audio processing functions such as noise cancellation or audio beamforming.

Abstract: An input multi-channel representation is converted into a different output multi-channel representation of a spatial audio signal, in that an intermediate representation of the spatial audio signal is derived, the intermediate representation having direction parameters indicating a direction of origin of a portion of the spatial audio signal; and in that the output multi-channel representation of the spatial audio signal is generated using the intermediate representation of the spatial audio signal.

Abstract: The quality of music output from audio systems is improved by simulating the effect of low frequency signals in the human ear. This thus allows listeners to perceive the lower frequency signals, even though the speakers may be incapable of providing such low frequency outputs. A method is provided for processing enhancing bass effect in audio signals. The method also results in the bass enhancement being computationally less intensive. The bass effect enhancement techniques are based on the response of sine and cosine transfer functions and on the directional independence of low frequency components. The human ear is unable to resolve directions from low frequency components. The bass effect enhancement technique alternatively is based on the response of an exponential transfer function.

Abstract: A method for processing audio signals optimizes the listening experience for hearing impaired listeners, as well as non-hearing impaired listeners, without forcing hearing impaired individuals to feel stigmatized by requiring them to employ special hearing-impaired equipment. A user actuated controller controls a mixture of a preferred audio signal and a remaining audio signal across a range sufficiently wide enough to encompass all individuals. The preferred audio is recorded and maintained separate from all remaining audio and delivered to the listener in a manner that maintains the separateness of the preferred audio and the remaining audio. The user actuated controller includes the capability of automatically maintaining the listener established ratio in the face of changes in the audio signal.

Abstract: A signal processing device includes a generation unit that generates a second signal from a first signal that is obtained by down mixing two signals; a mixing coefficient determination unit that determines, based on a value L and a value ?, a mixing degree for mixing the first signal and the second signal; and a mixing unit that mixes the first signal and the second signal based on the mixing degree determined by the mixing coefficient determination unit. The generation unit includes a first filter that generates a low frequency band signal in the second signal, from a low frequency band signal in the first signal; and a second filter that generates a high frequency band signal in the second signal, from a high frequency band signal in the first signal. The first filter is a filter unit which, for a complex-number signal, de-correlates an input signal and adds a reverberation component by using a delay unit and an all pass filter, and the processing unit is a filter unit different from the first filter.

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

Abstract: In an automatic correction process, automatic correction processing portions 60 are connected to a set reference channel and target channels, respectively, so that test signals will be input to the automatic correction processing portions 60, respectively. A rise detection portion 60a detects a rise in a test signal input to a corresponding channel, so that a value counted by a sample counter 61 is latched by a latch 60b at the rising timing of this test signal. In accordance with a difference between the counted value latched to the reference channel and the counted value latched to the target channel, a time difference is calculated. In accordance with the time difference, a delay time set for a channel delay means of the target channel is automatically corrected.

Abstract: Methods, systems and computer-readable medium reduce and/or eliminate errors in coding/decoding of streamed audio due to resetting of decoder state values based on playback buffer access. An inaudible compensation signal is included with the audio signal. The compensation signal is generated having a characteristic selected so that the encoded streamed audio signal substantially matches the reset state values at block boundaries. In an ADPCM example, the compensation signal is chosen such that the sum of the compensation signal and the original audio signal (=the compensated audio signal) has the characteristic that, at the block boundaries, the compensated audio signal matches the initial predictor value.

Abstract: The present invention relates to a vehicle communication system comprising a plurality of microphones adapted to detect speech signals of different vehicle passengers, a mixer combining the audio signal components of the different microphones to a resulting speech output signal, a weighting unit determining the weighting of the audio signal components for the resulting speech output signal, where the weighting unit determines the weighting of the signal components based upon non-acoustical information about the presence of a vehicle passenger.

Abstract: A method for operating a hearing device, an associated microphone system comprising at least two omnidirectional microphones, and a hearing device are provided. The microphones emit microphone signals and are electrically interconnected with one another in order to form directional characteristics. A damping of the upper frequency range of the microphone signals is determined from the lower frequency range of the microphone signals. The impression of a wide-band directional microphone is produced as a result.

Abstract: A system and method for flexibly outputting audio from a plurality of programs to the particular outputs that are desired for each audio stream. This routing flexibility enables the audio system to operate in a variety of configured usage modes, wherein each usage mode identifies the routing of each audio stream to one or more outputs such as SPDIF, HDMI, I2S, and DAC.

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

Abstract: In order to compensate tonal changes arising from a multi-path propagation of sound portions during the mixing of multi microphone audio recordings as far as possible it is suggested to form spectral values of respectively overlapping time frames of samples of each a first microphone signal (100) and a second microphone signal (101). The spectral values (300) of the first microphone signal (100) are distributed with formation of spectral values (311) of a first sum signal to the spectral values (301) of a second microphone signal (101) in a first summing level (310), whereat a dynamic correction of the spectral values (300, 301) of one of the two microphone signals (100, 101) occurs. Spectral values (399) of a result signal are formed out of the spectral values (311) of the first sum signal which are subject to an inverse Fourier-transformation and a block junction (FIG. 3).

Abstract: In an audio system, a control device includes a setting part that sets automatic compensation of each of a plurality of input ports of the input device into either ON state or OFF state and an adjusting part that adjusts an analog gain of each of the plurality of input ports. The input device includes the plurality of input ports, each input port including an amplifier that controls a level of an analog signal input to the input port based on the analog gain adjusted by the adjusting part, an AD converter that converts the analog signal from the amplifier into a digital signal, a compensator that controls a level of the digital signal from the AD converter based on a digital gain of the input port, and a selector that selects one of the digital signal from the AD converter and the digital signal from the compensator.

Abstract: In one embodiment the present invention includes a method for starting up a class D amplifier. The method comprises increasing, gating, and driving. The increasing includes increasing a duty cycle of a pulse train from a first duty cycle to a second duty cycle. The gating includes gating a signal based on the increasing of the duty cycle. The gating results in a gated signal. The driving includes driving an output signal from the gated signal to charge an output capacitor. The output capacitor is coupled to a speaker. The increasing of the duty cycle contributes to the charging of the output capacitor such that switching sounds detectable by the human ear are reduced.

Abstract: Audio signal processing apparatus is disclosed. The audio signal processing apparatus includes a first audio signal extracting section, a second audio signal extracting section, a sense-of-depth controlling section, a sense-of-sound-expansion controlling section, a control signal generating section, and a mixing section. The first audio signal extracting section extracts a main audio signal. The second audio signal extracting section extracts a sub audio signal. The sense-of-depth controlling section processes the extracted main audio signal to control a sense of depth. The sense-of-sound-expansion controlling section processes the extracted sub audio signal to vary a sense of sound expansion. The control signal generating section generates a first control signal with which the sense-of-depth controlling section is controlled and a second control signal with which the sense-of-sound-expansion controlling section is controlled.

Abstract: A sound device includes an input terminal for electronic signal input; a line out terminal adapted to output electronic signals input on the input terminal to an external area; a delay arrangement adapted to delay electronic signals input on the input terminal for a certain time; a main output arrangement adapted to output electronic signals delayed by the delay arrangement. The delay arrangement delays signals in order to accommodate a time difference between (1) electronic signals output from an external sound device connected with the line output terminal and (2) electronic signals output from the main output arrangement. The time difference between sound that is output from the sound device and the one that is output from an external sound device that is connected with line connection terminal can be eliminated. Thus, it can prevent sound quality decrease due to time difference from happening.

Abstract: The present invention relates to an apparatus and method for redeeming otherwise closed and concealed information contained in audio signals. This includes both the primary reference signal, and a plurality of redundant duplicate signals, substantially identical in all respects to the primary reference signal except in relation to magnitude and phase, for the purpose of unfolding, or opening the audio signal content into layers that result in an omni-directional acoustic signal, representing the sound as it would behave in nature. The audio reproduction system uses an in-phase circuit and a separate phase layering technique circuit to drive independent multiple mixed channels, to produce an open, substantially complete sound from a discrete audio signal, for the purpose of enabling a substantially complete audio signal to be formed, or to transform existing incomplete audio signals into a substantially complete audio signal.

Abstract: The present invention relates to an apparatus for processing an audio signal and method thereof.

Abstract: Once a mixer is set in a predetermined operation, an input-logical-channel selecting section supplies sound signals, input via a cascade input terminal, to an input signal processing section via an input patch section, so that the sound signals can be mixing-processed as sound signals of normal input channels. At the same time, a portion of sound signals input via an input terminal are supplied to mixing buses, so that these sound signals can be handled as cascade-related signals. In accordance with a model of another, or cascaded-to, mixer, arrangements are made such that normal-input/output-channel input terminals can be assigned to cascade input/output purposes. With a block diagram display section indicating what signals the individual input/output terminals are currently assigned to within the mixer, a user can grasp at a glance the current assignment state.

Abstract: The voice mixing method includes a first step for selecting voice information from a plurality of voice information, a second step for adding up all the selected voice information, a third step for obtaining a voice signal totaling the voice signals other than one voice signal, of the selected voice signals, a fourth step for encoding the voice information obtained in the second step, a fifth step for encoding the voice signal obtained in the third step, and a sixth step for copying the encoded information obtained in the fourth step into the encoded information in the fifth step.

Abstract: An apparatus and method for processing an audio signal are discussed. According to an embodiment, the apparatus includes a receiving unit receiving a downmix signal including plural objects, a bitstream including object information and level guide information; and a display screen displaying a graphical user interface including at least one level fader for receiving a user input for controlling level for at least one object among the plural objects. The graphical user interface displays a representation corresponding to the level guide information in association with the level fader.

Abstract: An audio mixing device includes: an adder that adds a plurality of PDM signals each converted from a plurality of digital audio signals; a D/A converter that performs D/A conversion on a digital audio signal outputted from the adder and outputs an analog audio signal; and a synchronization device that is provided prior to the adder, and that synchronizes each of a plurality of digital signals with one another by use of the same predetermined synchronization timing signal and outputs each of them to the adder.

Abstract: In an audio signal processing system, a plurality of audio signal processing apparatuses are previously linked to form a zone, and logically defined as devices belonging to a zone, and further the devices are grouped into a plurality of device groups within the zone. Each of the audio signal processing apparatuses includes a device group table that registers information specifying controls which are performed in respective audio signal processing apparatuses belonging to a device group in case that a group control command is given for collectively controlling the device group, and a master table that registers information specifying controls which are performed in respective device groups included in the zone in case that a zone control command is given for collectively controlling the zone. Both the collective control of the zone and the collective control of the device group are performed.

Abstract: The disclosure relates to sound playback and editing apparatus. The editing apparatus uses user interaction to allow the user to instinctively modify recorded sound. This can be achieved by converting a quality of the user's physical interactions with the editing apparatus into instructions for processing the sound. For example, in one embodiment the user can mix sound files by ‘mixing’, i.e. shaking, physical representations of those sound files (such as the recording medium on which the files are stored) alone or together.

Abstract: Methods and apparatuses for encoding and decoding a multi-channel audio signal are provided. In the encoding method, spatial information that is calculated based on a multi-channel audio signal and a downmix signal is encoded, and additional configuration information is generated based on information that is selected from the encoded spatial information. The downmix signal is encoded, and then, a bitstream is generated by combining the encoded downmix signal with the encoded spatial information. Thereafter, the additional configuration information is inserted into the bitstream. Therefore, it is possible to configure an optimum bitstream according to the circumstances by retransmitting all or part of information included in a header.

Abstract: A system for wirelessly controlling an audio mixer is disclosed. The system includes a controller disposed between audio input lines and a conventional audio mixer. The controller is wirelessly controlled via a remote computing device. The computing device may be a tablet, smartphone, laptop, personal computing device, or other computing device. Ideally, the computing device includes a graphic user interface and transmits signals to the controller via a wireless protocol such as the IEEE 802.15.1 standard protocol currently known as BLUETOOTH. Thus, a user may operate the wireless computing device while in an audience area or off stage area during a performance. The mixer may remain on stage with the performers, thus avoiding the need for wired connections leading from a stage area to an off stage area.

Abstract: Systems, methods, and computer program products for graphically displaying audio data are provided. In some implementations, a method is provided. The method includes displaying a graphical visual representation of digital audio data, the representation displaying a feature of the audio data on a feature axis with respect to time on a time axis. The method also includes receiving an input in the graphical visual representation selecting a range with respect to the feature and automatically extending the selected range with respect to time to define a selected region of the visual representation, where the extended range with respect to time is predefined and ignoring any component of the received input with respect to time.

Abstract: Systems, methods, and computer program products for editing digital audio data are provided. In some implementations a method is provided. The method includes receiving digital audio data incorporated in a plurality of audio tracks and receiving a selection of one or more audio tracks of the plurality of audio tracks for editing, the selected audio tracks being fewer than all of the plurality of audio tracks. The method also includes receiving an input specifying an audio effect to apply to the selected audio tracks and applying an inverse of the audio effect to one or more non-selected audio tracks in response to the input.

Abstract: An apparatus and method for autonomously mixing multiple devices capable of storing and playing audio signals is provided. Multiple devices can be mixed into one standard stereo signal that can then be played on any sort of powered speakers or amplifier. The apparatus is capable of receiving multiple audio inputs and can combine multiple iPods®, iPhones®, MP3 devices, or other devices capable of storing and playing audio signals, such that more than one device can be played at one time. No human intervention is required to control the device when the device is in autonomous mode. The autonomous mode can include random song playback using multiple devices.

Abstract: One or more attributes (e.g., pan, gain, etc.) associated with one or more objects (e.g., an instrument) of a stereo or multi-channel audio signal can be modified to provide remix capability. In some implementations, a method can include obtaining a first plural-channel audio signal having one or more objects; obtaining side information, at least some of which represents a relation between the first plural-channel audio signal and the one or more objects; obtaining a set of mix parameters; and generating a second plural-channel audio signal using the side information and the set of mix parameters.

Abstract: A slide manipulation device is mountable in a manipulator panel and includes a manipulator, a frame configured to slidably support the manipulator, and a circuit board. The frame has an opposing part facing the inside of the manipulation panel when the frame is mounted to the manipulation panel, the frame extending in a direction intersecting the opposing part and extending in a slide direction of the manipulator. The manipulator extends outward from the interior of the frame and is slidable in the slide direction to indicate a parameter for use in an electronic equipment. The circuit board is disposed at the outside of the frame along the extending direction of the frame, and is mounted to the frame for carrying out overall control to signalize a parameter set by the manipulator and to transmit the signalized parameter to the electronic equipment, and otherwise to receive the parameter from the electronic equipment and to slide the manipulator in response to the received parameter.

Abstract: In a digital mixer, a standard mode or a switched mode of an input patch is selectable. When shifting from the standard mode to the switched mode is selected, input port information in input patch data stored in a current memory is converted according to a port correspondence relation indicated by conversion data. When shifting from the switched mode to the standard mode is selected, the input port information in the input patch data stored in the current memory is reversely converted to original information according to the port correspondence relation indicated by the conversion data.

Abstract: A small compact easy to handle device with disc-jockey functionality is proposed, comprising a housing, loudspeaker system output means (23) for providing a loud-speaker sound output to at least one loudspeaker system, a display (13), means for 5 providing at least a first channel or at least a second channel as the loudspeaker sound output said computing device being characterized in that the housing is dimensioned to be handheld by a user and that it comprises—selection means (13) for selecting one of the first and second channel 10—control means (14, 18) comprising an input means for controlling at least one characteristic of the selected channel selected from the group of bend and play-back speed—cross-fading means (16) for cross-fading between the first and the second channel independently of the selected channel, said cross-fading means ena- 15 bling cross-fading by using one input means.

Abstract: A musical tone signal processing apparatus configured to extract musical tone signals that are signal processed for a plurality of localizations. Such an apparatus may be configured to carry out signal processing for signals that have been extracted by first retrieving processing (S100) and/or second retrieving processing (S200). The first retrieving processing (S100) and the second retrieving processing (S200) extracts a musical tone signal (e.g., the left channel signal and the right channel signal) that satisfies each of the conditions that have been set (e.g., frequency, localization, and maximum level) as the extraction signal. Accordingly, the extraction signal can be extracted to allow the musical tone signal processing apparatus to signal process the extraction signal for each of the plurality of conditions.

Abstract: A method and audio device are presented that preserve mono energy during downmixing of a hybrid coding process of an audio signal. The method includes calculating a stereo scaling factor in a group level that is definable within a stereo band. The method may also include updating the stereo scaling factor using an update rate and synchronizing the update rate of a spatial parameter during a fast changing transient portion of the signal. A number of groups in a first stereo band may be greater than a number of groups in a second stereo band, and the first stereo band may be a lower frequency band than the second band or may be perceptually more important than the second band.

Abstract: A system for facilitating access to multiple layer media items over communication network. The system comprises a media database which is used for storing multiple layer media items as independently accessible channels. The system further comprises plurality of subscriber applications which are connecting via a communication network, e.g. The Internet, to the media database. Users can use the subscriber application to access to each channel of the multiple layer media items independently.

Abstract: A loudness-compensating volume control method imposes a desired loudness scaling on an audio signal by processing the audio signal in both the digital and analog domains by receiving a desired loudness scaling, deriving a wideband gain component and one or more other gain components from the desired loudness scaling, applying in the digital domain modifications to the audio signal based on the one or more other gain components to produce a partly-modified audio signal, and applying in the analog domain modifications to the partly-modified audio signal based on the wideband gain component. Additional loudness modifications other than volume control loudness modifications on the audio signal may also be imposed.