Abstract: A method for encoding and decoding a digital audio signal is provided, said method comprising the steps of: encoding a first sequence of samples of the digital signal according to a transform encoding; encoding a second sequence of samples of the digital signal according to a predictive encoding; wherein the second sequence starts before the end of the first sequence, a subsequence common to the first and second sequences being thus encoded both by predictive encoding and by transform encoding.

Abstract: An electronic device with speakerphone and microphone is provided. The electronic device comprises a housing, a sound input unit, a sound output unit, an audio signal receiving terminal, an acoustic processing unit, a signal line and a sealing member. The housing has a microphone shell, a speaker shell and a soundproof structure. The soundproof structure is connected between the microphone shell and the speaker shell. The soundproof structure has a line output hole. The acoustic processing unit connects the sound input unit and the audio signal receiving terminal respectively to receive and mix two audio signals so as to generate an audio output signal. The audio output signal is transmitted to the sound output unit through a signal line. The sealing member is disposed around the line output hole and seals the line output hole.

Abstract: Managing audio of a device including transmitting a signal through a port of the device in response to an audio device coupling to the port, initializing an audio mixer of the device in response to detecting the signal, and configuring the audio mixer to manage audio transferred between the device and the audio device.

Abstract: In a mixer, each input channel adjustment unit includes a main system unit that generates a main system signal, and a plurality of MIX send units that generate a plurality of MIX send signals. In each MIX send unit, a signal selection unit selects an audio signal from a plurality of audio signals being involved in the main system unit and including at least an audio signal to be inputted to a filter unit of the main system unit and an audio signal to be inputted to a main system gain adjustment unit. A linked gain setting unit of the MIX send unit applies a gain that increases or decreases in linking with the gain of the main system gain adjustment unit, to the audio signal that is selected by the signal selection unit. A linked ON/OFF unit of the MIX send unit sets ON state or OFF state of the audio signal that is selected by the signal selection unit in linking with the ON state or OFF state of a main system ON/OFF unit.

Abstract: A method for controlling an audio output and a device supporting the same are provided. The method includes connecting an application operating device to an application output device through at least one communication channel, and transmitting audio information of at least one of a plurality of applications operating based on the at least one communication channel from the application operating device to the application output device.

Abstract: Efficiency for mixing audio signals from one or more portable user devices in an audio system employing a number of docking station is achieved by a docking station that includes a controller for detecting the presence of an activated mixer associated with another docking station in the system. The controller activates its associated mixer when no activated mixer associated with another docking host in the system is detected. When its associated mixer is activated, the controller in the one docking station causes its associated mixer to be connected both to the audio sound system for supplying an input signal thereto and indirectly to the output(s) of the one or more portable devices. Transfer of mixing from one docking station to another in the same system upon the occurrence of a particularly defined event is also described.

Abstract: Systems and methods for collectively generating sound are provided. A sound composition can include multiple component pieces and those component pieces can be divided amongst devices so that the devices can collectively generate the sound composition. For example, each component piece may be assigned to a different device and the each device may provide its assigned component pieces in a synchronized manner so that the overall sound composition is collectively generated. By generating a sound composition in this manner, the network may create new and exciting spatial effects.

Abstract: The present invention relates to a computer implemented system and method for generating an audio output file. The method including using one or more processors to perform steps of: receiving audio tracks, each audio track created according to audio parameters; separating each audio track into at least one selectable audio block, each audio block including audio content from a musical instrument involved in creating the audio track; assigning a unique identifier to each audio block; using the unique identifiers to select audio blocks, and generating the audio output by combining the audio blocks. The present invention prevents the use of the same combination of audio blocks in the generation of audio output to ensure that the audio output files generated a sufficiently unique. Also provided are audio file recording, editing and mixing modules enabling a user to have full creative control over mix and other parameters to modify as desired the audio file generated.

Abstract: An original loudness level of an audio signal is maintained for a mobile device while maintaining sound quality as good as possible and protecting the loudspeaker used in the mobile device. The loudness of an audio (e.g., speech) signal may be maximized while controlling the excursion of the diaphragm of the loudspeaker (in a mobile device) to stay within the allowed range. In an implementation, the peak excursion is predicted (e.g., estimated) using the input signal and an excursion transfer function. The signal may then be modified to limit the excursion and to maximize loudness.

Abstract: A method for providing a user with mixed content for a device. The method includes pre-rendering mixed content comprising a mix of a first source track and a second source track identified as being compatible with the first source track. The pre-rendered content is stored in a database. Upon receiving a user request which identifies the first source track as a track to be included in the mixed content for a device, the system retrieves a list of second source tracks having characteristics compatible with the first source track and provides the list to the user. Then, the system receives request identifying the second source track as a track to be included in the mixed content along with the first track. In response, the system retrieves the pre-rendered mixed content from the database and sends the pre-rendered mixed content to the device.

Abstract: A mixer includes a plurality of input channels, a plurality of input groups, and a plurality of group strips that adjust values of parameters of the input groups. The mixer preliminarily prepares a plurality of group presets having data required for setup of group modules as presets, and each group preset has attribute information including musical genre, number of input channels, and instrument kind. By selecting a musical genre, a user can change the group presets which are being applied at present to the plurality of input groups to other group presets corresponding to another musical genre while keeping the number of input channels and the instrument kinds which are being used at present between before and after the change of genre.

Abstract: A method including determining location of at least one second device relative to a first device, where at least two of the devices are configured to play audio sounds based upon audio signals; and mixing at least two of the audio signals based, at least partially, upon the determined location(s).

Abstract: Systems and methods for wirelessly recording multi-track audio files without the data corruption or loss of data that typically occurs with wireless data transmission. In some aspects, each performer is equipped with a local audio device capable of locally recording the respective performer's audio while also transmitting it to a master recorder. Functions of the local audio device may be adjusted remotely. The locally recorded audio may be used to repair or replace any audio lost or corrupted during transmission to the master recorder. Such repair or replacement may be performed electronically or via playback of the locally recorded audio. In other aspects, a master recorder is not required since all locally recorded audio may be combined or otherwise processed post-recording. Locally recorded audio may include identifiers to aid in post-recording identification of such audio. A multi-memory unit is also provided to facilitate manipulation and processing of audio files.

Abstract: A wireless cellular telephone with an audio codec for converting digital audio signals to analog audio signals. The audio codec comprises two digital audio bus interfaces for coupling to respective digital audio buses, and a digital-only signal path between the two digital audio bus interfaces, such that no analog processing of the audio signals occurs in the digital-only signal path.

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

Abstract: 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: An audio signal processing device and a parameter adjusting method, configured such that: a two-channel screen displaying a plurality of parameters for each of two channels allocated to any two channel strips, in accordance with operation of edit buttons of the two channel strips, is displayed on a display; each of a plurality of parameters for one channel, out of the two channels displayed in the two-channel screen, are allocated to a control of each channel strip in a first group; each of a plurality of parameters for the other channel are allocated to a control of each channel strip in a second group; and values of the parameters allocated to the controls are adjusted in accordance with operation of the controls.

Abstract: A method of speech enhancement in a room (10) includes the steps of capturing audio signals from a speaker's voice by a microphone (12), estimating an ambient noise level in the room from the captured audio signals, processing the captured audio signals by an audio signal processing unit (20), estimating a reverberation level, determining the gain to be applied to the captured audio signals by the audio signal processing unit according to a comparison between the estimated ambient noise level and the estimated reverberation level, and generating sound according to the processed audio signals by a loudspeaker arrangement (24) located in the room, wherein the reverberation level is the level of reverberant components of the sound generated by the loudspeaker arrangement.

Abstract: A mixing system provides both clipping protection and signal level conservation while the system operates in the original width type. The mixing system includes a first input multiplier multiplying a first digital input signal by a first gain value to provide a first scaled signal, a second input multiplier multiplying a second digital input signal by the first gain value to provide a second scaled signal, a combiner combining the first scaled signal and the second scaled signal to provide a combined signal, a soft limiter soft limiting the combined signal by reducing some of the amplitudes of the combined signal to provide a soft limited signal, and an output multiplier multiplying the soft limited signal by a second gain value to provide a mixed output signal, wherein the first gain value is a value that is equal to an inverse value of the second gain value.

Abstract: An apparatus for generating at least one audio output signal representing a superposition of at least two different audio objects comprises a processor for processing an audio input signal to provide an object representation of the audio input signal, where this object representation can be generated by a parametrically guided approximation of original objects using an object downmix signal. An object manipulator individually manipulates objects using audio object based metadata referring to the individual audio objects to obtain manipulated audio objects. The manipulated audio objects are mixed using an object mixer for finally obtaining an audio output signal having one or several channel signals depending on a specific rendering setup.

Abstract: An apparatus for generating an audio output signal having two or more audio output channels from an audio input signal having two or more audio input channels includes a provider and a signal processor. The provider is adapted to provide first covariance properties of the audio input signal. The signal processor is adapted to generate the audio output signal by applying a mixing rule on at least two of the two or more audio input channels. The signal processor is configured to determine the mixing rule based on the first covariance properties of the audio input signal and based on second covariance properties of the audio output signal, the second covariance properties being different from the first covariance properties.

Abstract: Techniques are disclosed relating to generating music content. In one embodiment, a method includes determining one or more musical attributes based on external data and generating music content based on the one or more musical attributes. Generating the music content may include selecting from stored sound loops or tracks and/or generating new tracks based on the musical attributes. Selected or generated sound loops or tracks may be layered to generate the music content. Musical attributes may be determined in some embodiments based on user input (e.g., indicating a desired energy level), environment information, and/or user behavior information. Artists may upload tracks, in some embodiments, and be compensated based on usage of their tracks in generating music content. In some embodiments, a method includes generating sound and/or light control information based on the musical attributes.

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: A mixing device includes a signal data generation unit that generates signal data on a time-frequency plane composed of a time-axis and a frequency-axis from each of two or more input signals derived from a time domain, a signal processing unit that receives the signal data for the two or more input signals, and performs a mixing process that includes an adding operation of the signal data for each corresponding point on the time-frequency plane of the signal data for each of the input signals, and an output signal generation unit that receives the operation result of the signal processing unit and outputs an output signal formed by converting the inputted signal into a signal of a predetermined format.

Abstract: A speaker unit used for a musical instrument and configured to generate music sounds based on a performance operation in a main body of the musical instrument, including: an electrostatic speaker; and a stationary portion which is a portion fixed to the main body of the musical instrument or which is a member attachable to the main body of the musical instrument, wherein the electrostatic speaker is supported by the stationary portion such that at least one of a position and a posture of the electrostatic speaker with respect to the stationary portion is variable.

Abstract: Processor-implemented methods and systems for visually-assisted mixing of audio using a spectral analyzer are disclosed. The system calculates and displays a spectral view for each track in an arrangement in a multi-track view. A user can then request modification of the spectral view. In response to this request for modification, the system automatically adjusts associated mixing parameters for the modified track so that the spectral output of the track substantially matches the user-requested modified spectral view. This allows a user to visually mix an arrangement by imputing a desired spectral result and having the program make the changes necessary to achieve it.

Abstract: Our device relates to an audio signal processing device, a means for sharing expression control signals between physically separate sound processing devices, a method of deriving related expression control signals from a common expression control signal, and a means for mapping control signals from an expression device to one or more of the various control parameters of the audio signal processing device.

Abstract: An audio mixing method, apparatus and system, which can ensure sound quality after audio mixing and reduce consumption of computing resources. The method includes: receiving an audio stream of each site, and analyzing the audio stream of each site to obtain a sound characteristic value of a sound source object; selecting, according to a descending sequence of sound characteristic values of sound source objects, a predetermined number of sound source objects from the sound source objects to serve as main sound source objects; determining, according to a relationship between a target site and the sites where the main sound source objects are located, audio streams that require audio mixing for the target site; and performing audio mixing on the audio streams that require audio mixing for the target site and sending the audio streams after the audio mixing to the target site.

Abstract: A method for applying audio effects to one or more tracks off musical compilation is disclosed. The method comprises identifying a first virtual musician to be associated with a first instrument track; identifying a first virtual producer to be associated with the first instrument track; applying a first series of effects to the first instrument track, the first series of effects being dependent upon the first virtual musician; and applying a second series of effects to the first instrument track, the second series of effects being dependent upon the first virtual producer.

Abstract: A volume level and crossfade controller with programmable fades and audio stream priority controls saturation for a multiple audio stream mixer. Start of fade and end of fade events trigger updates of relative volume level targets and volume ramps.

Abstract: An apparatus for controlling a target device including a first input device configured to provide a first input to the target device, a second input device configured to provide a second input to the target device, and a control mixer configured to generate an output using a policy, the first input and the second input, wherein the output comprises a feedback and a target device output, wherein the feedback comprises some function of the state of the target device, the policy, and the state of each input device connected to the control mixer.

Abstract: There is described Audio driver circuit comprising a main, 1-bit digital-to-analog converter, adapted to receive a 1-bit oversampled digital audio signal. Thanks to an auxiliary digital-to-analog converter which converts into the analog domain a digital offset compensation value stored in e.g. an OTP memory register 34, and to a summing and filtering arrangement, it is possible to reduce the offset cancellation granularity in order to compensate for the offset of the main digital-to-analog converter in the output signal.

Abstract: A method of managing a power state of devices in a home entertainment system using a wireless home entertainment hub comprises obtaining a registration of a plurality of devices, where the plurality of devices includes at least one sink device and at least one source device. A selection of one of the source devices is received for presentation of program content to a user in the home entertainment system. A subset of the at least one sink device is determined for presentation of the program content to the user based on the selection of the source device. Each sink device in the subset of the at least one sink device is ensured to be active.

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

Abstract: Provided are an audio encoding method and apparatus and an audio decoding method and apparatus in which audio signals can be encoded or decoded so that sound images can be localized at any desired position for each object audio signal. The audio decoding method generating a third downmix signal by combining a first downmix signal extracted from a first audio signal and a second downmix signal extracted from a second audio signal; generating third object-based side information by combining first object-based side information extracted from the first audio signal and second object-based side information extracted from the second audio signal; converting the third object-based side information into channel-based side information; and generating a multi-channel audio signal using the third downmix signal and the channel-based side information.

Abstract: An electrostatic transducer loudspeaker includes a filterless class-D amplifier to modulate an audio input signal to generate a modulated signal containing a PWM switching carrier component, and a transformer directly connected at an output side of the filterless class-D amplifier and directly connected at an input side of an electrostatic transducer, whereby the equivalent capacitance of the electrostatic transducer and the equivalent inductance of the transformer establish a resonance circuit to demodulate the modulated signal to generate an AC voltage to drive the electrostatic transducer.

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: Once a human operator operates a fader knob of a desired channel downwardly toward a predetermined position, resistance force against the fader knob is generated. As the human operator further lowers the fader knob beyond another predetermined position against the resistance force, the channel is set in a CUE-ON state, so that an audio signal (pre-fader signal) of the channel can be test-listened to as a CUE signal. Then, once the human operator operates the fader knob upwardly, the CUE-ON state of the signal is canceled, so that a sound volume level of the channel can be controlled in accordance with an operating position of the fader knob.

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

Abstract: A controller remote-controlling a digital mixer which performs signal processing to an input signal by a DSP to output the processed signal is provided with functions of: accepting the setting of level of a dummy signal; calculating a gain of the signal processing at each stage in the DSP based on a value of a parameter used for the remote controlling; calculating level that the dummy signal would have at a reference point selected by a reference point selection button if the dummy signal is assumed to be inputted to the DSP, based on the level of the dummy signal and the calculated gain; and displaying the calculated level in a level display portion.

Abstract: An audio signal processing device includes multiple input reception units to which analog audio signals, on which watermark information indicating identification information is superimposed, are input, an extraction unit that is adapted to extract the identification information from each of the analog audio signals input to the multiple input reception units, and a display unit for performing display depending on the identification information extracted by the extraction unit in correspondence with the input reception unit to which the analog audio signal, from which the identification information is extracted, is input, or signal processing unit for performing signal processing depending on the identification information extracted by the extraction unit for the analog audio signal, from which the relevant identification information is extracted, and outputting the processed analog audio signal.

Abstract: One example embodiment increases perceived signal strength of a sound signal based on persistence of hearing. When the ear perceives a signal it takes a finite length of time to process it. During that time period the ear does not recognize input and is, in effect, “Turned Off”. In accordance with one aspect of the technology, these periods of “unheard” audio input are utilized so that they supplement the information which the ear sends to the brain. An exemplary system includes a switch, configured to route the audio alternatively to two signal paths—one which includes a delay circuit that delays that audio by an amount equal to the ear's persistence of hearing interval. The system also includes a signal combiner configured to combine outputs from the two signal paths, so as to provide the brain with a signal that it will perceive as being twice as loud as the original.

Abstract: A system and method for enhancing audio, the method including receiving audio input tracks, with at least one audio input track including a restriction parameter, determining a restricted audio input track, where the restricted audio input track is the audio input track including the restriction parameter, manipulating another audio input track based on musical properties of the restricted audio input track, and combining the restricted audio input track and the manipulated audio input track into a single output audio track.

Abstract: A interface circuit for an acoustic transducer provided with a first detection structure and a second detection structure has: a first input and a second input; a first processing path and a second processing path coupled, respectively, to the first input and second input and supply a first processed signal and a second processed signal; and a recombination stage, which supplies a mixed signal by combining the first processed signal and the second processed signal with a respective weight that is a function of a first level value of the first processed signal. The first and second inputs receive a respective detection signal associated, respectively, to the first detection structure and to the second detection structure of the acoustic transducer; and an output stage the first processed signal, the second processed signal or the mixed signal, on the basis of a second level value of the first processed signal.

Abstract: A volume level and crossfade controller with programmable fades and audio stream priority controls saturation for a multiple audio stream mixer. Start of fade and end of fade events trigger updates of relative volume level targets and volume ramps.

Abstract: A method and apparatus for processing an audio signal is disclosed. Herein, the method includes receiving a downmix information having at least one independent object and a background object downmixed therein; receiving an object information and a mix information; and extracting at least one independent object from the downmix information using the object information and the enhanced object information.

Abstract: A controller includes a plurality of channel strips to which are allocated any of a plurality of channels to be handled in a DAW. Each of the channel strips includes a slide operation reception section that receives a slide operation. The slide operation reception sections of the plurality of channel strips are constructed to function as a single slide operation reception unit and to receive a slide operation performed in an arranged direction of the channel strips. Thus, by the slide operation, a user can intuitively and collectively change the allocated channels of the strip channels with a feeling as if merely changing (shifting) the allocated channel of a given channel strip to another channel strip.

Abstract: An apparatus for extracting a signal from convolutive mixtures includes a receiving unit which includes two or more receivers and receives a signal; a transfer function calculation unit which calculates transfer functions for demixing; and a demixing unit which demixes the received signal using the calculated transfer functions. The transfer function is determined such that a signal is extracted from a source closest to the receivers, and is calculated on the basis of a transfer function for a path to each receiver being approximated to a delta function as closer to the source.

Abstract: A method and apparatus for processing an audio signal is disclosed. Herein, the method includes receiving a downmix information having at least two independent objects and a background object downmixed therein; separating the downmix information into a first independent object and a temporary background object using a first enhanced object information; and extracting a second independent object from the temporary background object using a second enhanced object information.

Abstract: The audio production console includes a touch-sensitive display cradled by a chassis that also includes a console panel and meter bridge. The touch-sensitive display includes an IR sensor perimeter and related circuitry for processing multi-touch events simultaneously and translating those events into parameter controls. The parameter controls are represented digitally on the display in a GUI reminiscent of traditional analog production console layouts. Software tracks movement of the digital parameter controls and works in conjunction with hardware to manipulate production audio as one or more on-screen parameters are adjusted by touching the display.