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: 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: 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 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: 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: 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 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: 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: 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 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 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 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 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 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 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.

Abstract: Systems and methods may provide for capturing one or more inbound wireless transmissions and identifying a remote user movement based on at least one of the one or more inbound wireless transmissions. Additionally, a local haptic output may be generated, by an actuator, based on the remote user movement. In one example, the actuator is a piezoelectric actuator.

Abstract: An audio signal processing method and an audio signal processing apparatus for synchronizing audio based on synchronization error between audio signals.

Abstract: An audio or video conferencing system having an array comprised of two or more microphones and having one or more loudspeakers positioned proximate to the microphones. The conferencing system also has acoustic echo cancellation (AEC) functionality that removes direct and reverberant acoustic echo from microphone signals in two separate acoustic echo cancellation operations. The first AEC operation removes direct acoustic echo prior to a beamforming operation, and the second AEC operation removes reverberant acoustic echo from a microphone signal subsequent to the beamforming operation.

Abstract: An article of functional headwear is provided. The article of headwear including a material configured for surrounding at least a portion of an individual's head, an audio delivery device movably positioned within the material, and a microprocessor positioned within the material, the microprocessor being coupled to the audio delivery device.

Abstract: A method, apparatus and computer program product are provided to facilitate user interaction with, such as modification of, respective audio objects. An example method may include causing a multimedia file to be presented that includes at least two images. The images are configured to provide animation associated with respective audio objects and representative of a direction of the respective audio objects. The method may also include receiving user input in relation to an animation associated with an audio object or the direction of the audio object represented by an animation. The method may further include causing replay of the audio object for which the user input was received to be modified.

Abstract: A mixer includes a plurality of channel strips to which are variably allocated respective objects of operation, and a memory of the mixer stores layer data comprising information that, for each of the channel strips, designates a channel or a channel group as an object of operation of the channel strip, or designates the channel strip as a deploying channel strip for individually deploying thereto any one of the channels belonging to a given group. When objects of operation are to be allocated to the individual channel strips, no channel or group is allocated to each deploying channel strip, and a setting is made to the effect that the channel strip is to be used for a channel deploying purpose. In response to a deploying instruction of a given group, individual channels belonging to the given group are deployed to the deploying channel strips.

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: An audio signal in a plurality of input channels each including an equalizer and a dynamics as a first signal processing part is taken out as a cue signal by a cue switch 126, and the cue signal is outputted to a monitoring output terminal via a cue output channel 160. In the cue output channel, there are provided an equalizer 163 and a dynamics 164, as a second signal processing part, which perform signal processing common to the first signal processing part of the input channel in accordance with values of second parameters which are editable independently from first parameters used by the first signal processing part. In this configuration, it is preferable that the values of the second parameters may be reflected onto signal processing in the input channel in accordance with an instruction by a user.

Abstract: A vehicle and a control method of the vehicle are provided. The vehicle includes a first controller that is configured to provide a media sound source via a first virtual speaker. Additionally, the apparatus includes a second controller that is configured to provide the event sound source via a second virtual speaker, which is formed in a layer different from the first virtual speaker, when an event sound source is input.

Abstract: A multiband ducker is configured to duck a specific range of frequencies within a music signal in proportion to a corresponding range of frequencies within a speech signal, and then combine the ducked music signal with the speech signal for output to a user. In doing so, the multiband ducker separates the music signal into different frequency ranges, which may include low, middle, and high-range frequencies. The multiband ducker then reduces the amplitude of the specific range of frequencies found in the speech signal, typically the mid-range frequencies. When the ducked music signal and the speech signal are combined, the resultant signal includes important frequencies of the original music signal, including low-range and high-range, thereby allowing perception of the music signal to continue in relatively uninterrupted fashion. Additionally, the combined signal also includes the speech signal, allowing for the perception of intelligible speech concordant with the perception of music.

Abstract: A device for generating a multitude of loudspeaker signals based on a virtual source object which has a source signal and a meta information determining a position or type of the virtual source object. The device has a modifier configured to time-varyingly modify the meta information. In addition, the device has a renderer configured to transfer the virtual source object and the modified meta information to form a multitude of loudspeaker signals.

Abstract: In one aspect, the present application is directed to a device for providing different levels of sound quality in an audio entertainment system. The device includes a speech enhancement system with a reference signal modification unit and a plurality of acoustic echo cancellation filters. Each acoustic echo cancellation filter is coupled to a playback channel. The device includes an audio playback system with loudspeakers. Each loudspeaker is coupled to a playback channel. At least one of the speech enhancement system and the audio playback system operates according to a full sound quality mode and a reduced sound quality mode. In the full sound quality mode, all of the playback channels contain non-zero output signals. In the reduced sound quality mode, a first subset of the playback channels contains non-zero output signals and a second subset of the playback channels contains zero output signals.

Abstract: A headset audio system is provided, comprising at least a headset, a control unit, and a control assignment selector, wherein the headset is adapted to provide output audio to a user from at least a first and a second audio signal and the control unit is provided to allow user adjustment of one or more audio parameters and comprises at least a first assignable audio control and a second assignable audio control.

Abstract: A management software process establishes management sessions with DAW processes and controllers on a network. Further, control sessions are established between the DAW processes and controllers, respectively. Thus, the control sessions between DAW processes and controllers included in one music system can be managed together by the management software process via a dedicated management session.

Abstract: When recalling a preliminary stored scene including parameter values regarding a plurality of input/output ports respectively receiving or outputting an audio signal with respect to outside, a plurality of transmission/reception ports respectively transmitting or receiving an audio signal with respect to another equipment via a network, a signal processing channel processing an audio signal, and a patch transmitting an audio signal among the plurality of input/output ports, the plurality of transmission/reception ports and the signal processing channel, a setting to leave parameter values defining connection between the input/output ports and the transmission/reception ports by the patch before the recall as they are can be performed by a particular group safe setting button.

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

Abstract: Embodiments are provided for controlling playback volumes of a group of one or more playback zones in a network media system via a user interface provided on a touch screen display. For instance, touch inputs may be provided to the user interface via the touch screen to move a volume indicator along a graphical representation of a volume scale on the user interface to adjust a particular playback volume level of a playback zone or the group of the one or more playback zones. Further, touch inputs such as a dwell touch input or a swipe touch input may be configured to cause the particular playback volume level to be applied to other playback zones in the network media system, such that each playback zone the playback volume level is applied to has a respective playback volume level matching, or substantially matching the particular playback volume level.

Abstract: To a first analog signal that has been inputted, a first channel effect is added by an analog process that can be controlled with controls provided in a first channel strip. To a second analog signal that has been inputted, a second channel effect is added by a digital process, the second channel effect being controlled by a fourth control provided in a second channel strip. The second channel effect is preferably performed by a digital effect DSP.