Abstract: Disclosed is a source driver for a display apparatus which is insensitive to power noise, and a configuration of filtering an influence of power noise, which is introduced from an exterior of the source driver or occurs in an interior thereof, to an operation of the source driver. The present invention is applied to the case of receiving a clock signal and a data signal through the single signal line, and is embodied such that a source driver for driving a display apparatus for achieving a high speed operation and a large screen has a characteristic insensitive to power noise.

Abstract: A circuit for common electrode voltage generation includes: a VCOM driver configured to output alternating voltage levels at an output thereof, the output being connected to a display panel; a switching circuit with a plurality of inputs and an output, being configured to select one of voltage levels at the inputs at a time and thereby to output alternating voltages levels at the output of the switching circuit; and a stabilizing capacitor with one end connected to the output of the VCOM driver, and the other end connected to the output of the switching circuit.

Abstract: A method for the distribution of audio and visual media includes: receiving at least one visual content item to be displayed, and an audio content item and at least one trigger condition for each of the at least one visual content item; storing, in a database, the received at least one visual content item and corresponding audio content item and at least one trigger condition; identifying at least one display condition of a plurality of display conditions; identifying, in the database, a specific visual content item, wherein the at least one trigger condition corresponding to the specific visual content item is met based on the identified at least one display condition; displaying, by a light projection display device, the identified specific visual content item; and wirelessly transmitting the audio content item corresponding to the specific visual content item for audible emission by a mobile communication device.

Abstract: A semiconductor device having a normal function means is provided, in which the amplitude of an output signal is prevented from being decreased even when a digital circuit using transistors having one conductivity is employed. By turning OFF a diode-connected transistor 101, the gate terminal of a first transistor 102 is brought into a floating state. At this time, the first transistor 102 is ON and its gate-source voltage is stored in a capacitor. Then, when a potential at the source terminal of the first transistor 102 is increased, a potential at the gate terminal of the first transistor 102 is increased as well by bootstrap effect. As a result, the amplitude of an output signal is prevented from being decreased.

Abstract: In a method and devices, data for the display of an application on a screen is provided. In addition, information with regard to a degree of distraction of the application, e.g., information regarding user-interface elements, and/or information regarding a degree of distraction of a driver of a vehicle (110) is provided. As a function of this information and a driving situation of the vehicle, it is decided whether the application will be displayed on the screen.

Abstract: Systems and methods for presenting two-dimensional (2D) content within a virtual reality (VR) presentation are disclosed. Some embodiments include providing the VR presentation to a user, receiving information of the 2D content for display during the VR presentation from an external source and obtaining location information related to a display location within the VR presentation for displaying the 2D content. The 2D content may be displayed at the display location based on the received information and the location information, wherein the display location corresponds to a 2D display rendered within the VR presentation.

Abstract: In a display device, a timing controller is configured to transfer first display data and part of second display data to a first source driver via a first data transfer line at a high transfer frequency, and to transfer the remaining part of the second display data to a second source driver via the first data transfer line at a low transfer frequency, and the first source driver is configured to transfer the received part of the second display data to the second source driver via a second data transfer line.

Abstract: Systems and methods for self-adjusting displays are provided. A self-adjusting display may receive a display characteristics shift profile, where the display characteristics shift profile can be generated based on calibration event data collected from a plurality of displays. The display characteristics shift profile can be utilized as an estimate or prediction of a shift in display characteristics the self-adjusting display may experience over time. The self-adjusting display may calculate adjustment factors or curves to compensate for the estimated/predicted shift in display characteristics to lessen or altogether eliminate the need for performing actual calibration on the self-adjusting display.

Abstract: Circuits for programming a circuit with decreased programming time are provided. Such circuits include a storage device such as a capacitor for storing display information and for ensuring a driving device such as a driving transistor drives a light emitting device according to the display information. The present disclosure provides driving schemes for decreasing flickering perceived while displaying video content by introducing idle phases in between in emission phases to increase the effective refresh rate of a display. Driving schemes are also disclosed for reducing the effects of cross-talk by ensuring that programming information is refreshed in a display array that utilizes a driver connected to multiple data lines via a multiplexer.

Abstract: An imaging and display apparatus for passive displays evaluates the illumination of an input scene and incorporates data representative of such input scene within a transfer media. The transfer media may be a broadcast or transmission of image data, illumination data and gamma information that can be received by a display system, which includes a passive display illuminated by incident light, to display images while adjusting the incident light and/or gamma based on the received data to reduce energy requirements, to increase contrast or shades of gray in the displayed image, and to optimize light source operation for color fidelity. The data may be provided as a video signal, modulated video signal, s-video signal, digital signal, or other signal that can be used by a passive display system to display images.

Abstract: There are disclosed a backlight source driving circuit and a display apparatus. The backlight source driving circuit comprises a switch field effect transistor (Q1), a freewheeling field effect transistor (Q2), a power supplying module, a control module, a LED group (Z) and a feedback module, the power supplying module is configured to supply power to the LED group (Z), the control module is configured to control turn-on timings of the switch field effect transistor (Q1) and the freewheeling field effect transistor (Q2) to be opposite, and the feedback module is configured to provide a feedback voltage to the control module. The backlight source driving circuit is disposed in the display apparatus. By utilizing the field effect transistor as the freewheeling device in the backlight source driving circuit, the power consumption of the driving circuit is decreased effectively and an efficiency of the driving circuit is enhanced.

Abstract: A method, apparatus and computer program product are provided for multi-resolution point of interest boundary identification in digital map rendering. A method is provided for receiving a point of interest selection indication. The method also includes receiving point of interest boundary data and map data associated with the selected point of interest from a memory. The boundary data is based on the physical shape of the structure or region associated with the point of interest. The method also includes overlaying point of interest boundary data on the map data; and causing the map data with point of interest boundary data overlay to be displayed on a user interface.

Abstract: Embodiments of a display apparatus of an energy management system are presented. The display apparatus may include: a first screen updating unit including a drawing unit configured to determine a screen to be output based on a screen operation input and update data and an object included in the determined screen to configure an output-scheduled screen and a first temporary storage unit configured to store the configured output-scheduled screen; and a second screen updating unit configured to receive the stored output-scheduled screen and output the received output-scheduled screen through a screen output unit.

Abstract: An aspect of the present invention proposes a novel approach that can reduce the total number of the overlays to be composited during the display of graphical output in a mobile computing device. As a result, the total number of memory bandwidth and the usage of a graphics processing unit by a pre-compositor can be decreased significantly. According to one embodiment, this new approach is implemented with a display panel with embedded memory which supports a partial update, or refresh feature. Which such a feature, the layer compositor (typically either the display controller or GPU) is able to keep track of actively updating regions of a display panel by checking if each layer has new content to be displayed.

Abstract: A modular apparatus and method of varying the playability and tonal characteristics of a stringed instrument such as an acoustic or electric guitar which is commonly provided with a body and a fretboard to the instrument's neck. The modular apparatus includes a longitudinal rail, accessible fasteners, intermediary spacer blocks and a plurality of adjustable and removable frets. The rail is configured to connect with the neck of the stringed instrument and to provide a base for the frets. The frets are configured to move as needed within apertures within the rail. The frets are also configured to receive and support the instrument's strings. The frets are selectively adjustable relative to the rail by the user to change the tonal and playability characteristics of the instrument.

Abstract: A drum beater shaft with a flexible coupling means disposed between the base portion of the shaft and the head portion of the shaft is disclosed. The base portion of the shaft is fixed and hingedly attached, typically, to a pedal operated actuator. The head portion of the shaft contains a striking device used to contact the drum when actuated by the pedal. In one preferred embodiment, the flexible coupling means disposed between the base portion and head portion of the drum beater shaft takes the form of a spring surrounded by an elastic sheath. In another preferred embodiment, the flexible coupling means takes the form of a universal joint surrounded by an elastic sheath, In yet another preferred embodiment, the flexible coupling means takes the form of a ball an socket joint surrounded by and elastic sheath.

Abstract: Improvements in a tilting, lifting stand are presented. The tilt, lift stand is for a musical instrument stand or to elevate/tilt a speaker and or amplifier and rigidly supports a sousaphone brass type instruments, speaker or amplifier in an in a stored and in a playable or elevated position. The stand supports all or the majority of the weight of the instrument or speaker or amplifier in a static position to relieve the weight of the musical instrument or speaker or amplifier. The stand has legs that are extendable to rest on multiple levels as might be found in a stadium. The stand further has a securing mechanism that gently grasps a tubular section to hold the musical instrument. The speaker stand can be easily collapsed for transportation and storage.

Abstract: A method and system for analyzing patterns in the relationships of notes of an input piece of music. The method comprises generating a set of the most frequently occurring note pitches in ascending pitch order that matches an interval pattern, and detecting out-of-key pitches that lie outside of this interval pattern. One or more potential key sequence bifurcations are identified which represent a list of possible key sequences according to forwards and backwards analysis. By finding patterns of repetition in the chordal sequences that may be generated according to these key sequence bifurcations, a key sequence that allows the most frequently recurring chord sequences may be chosen. Chord sequences may be analyzed by using ghost chords, temporary harmonic structures that are created, updated and finalized over time according to a combination of essential and inessential note fragments. The method further comprises identifying non-harmony pitches according to the analyzed chord sequence.

Abstract: The present document relates to audio coding systems which make use of a harmonic transposition method for high frequency reconstruction (HFR), and to digital effect processors, e.g. so-called exciters, where generation of harmonic distortion adds brightness to the processed signal. In particular, a system configured to generate a high frequency component of a signal from a low frequency component of the signal is described. The system may comprise an analysis filter bank (501) configured to provide a set of analysis subband signals from the low frequency component of the signal; wherein the set of analysis subband signals comprises at least two analysis subband signals; wherein the analysis filter bank (501) has a frequency resolution of ?f.

Abstract: Techniques are described that relate to various aspects of converting the mechanical energy of instrument strings to digital representations for use in a variety of applications.

Abstract: The present invention relates to a sound absorbing and insulating material with improved heat resistance and moldability and a method for manufacturing the same, more particularly to a sound absorbing and insulating material having, as a surface layer, a heat-resistant material prepared by impregnating a binder into a nonwoven fabric formed of a heat-resistant fiber stacked on one side of a base layer formed of a conventional sound absorbing and insulating material, and a method for manufacturing the same. The sound absorbing and insulating material of the present invention is a conventional sound absorbing and insulating material has improved sound-absorbing property, flame retardancy, heat-insulating property and heat resistance as compared to the conventional sound absorbing and insulating material, is applicable to parts maintained at high temperatures of 200° C. or higher due to the surface layer and is moldable into a desired shape during the curing of the binder impregnated into the surface layer.

Abstract: A headphone includes an electroacoustic transducer and a support structure for suspending the transducer adjacent to a user's ear when worn by the user such that the headphone is acoustically open. A first microphone is coupled to one or more of the transducer and the support structure such that the first microphone is located in a substantially broadband acoustic null of the transducer. A processor is coupled to the headphone. The microphone receives sound pressure waves and outputs a related electronic signal to the processor. The processor uses the electronic signal to operate the transducer to reduce targeted sound pressure waves at the user's ear.

Abstract: Stability is provided in an active noise reduction (ANR) headphone by measuring a sound field to generate an input signal, filtering and applying a variable gain to the input signal to produce a first filtered signal using a first filter and a variable gain amplifier in an ANR signal pathway, outputting the filtered signal, and simultaneously with outputting the first filtered signal, sampling a signal at a point in the ANR signal pathway and filtering the sampled signal using a second filter to produce a second filtered signal. The second filtered signal is compared to a threshold, and if the comparison finds that the second filtered signal is greater than the threshold signal, the gain of the variable gain amplifier is changed to attenuate the first filtered signal. The second filter applies different gains, different by at least 10 dB, in different frequency ranges between 10 Hz and 10 kHz.

Abstract: A sound system for a motor vehicle exhaust gas system for emitting emulated engine sounds, including at least one loudspeaker and at least one housing accommodating at least one loudspeaker. The housing is mounted separately from a motor vehicle exhaust gas system in terms of the routing of an exhaust gas flow, at least one sound guidance tube having an outlet zone being mounted on the housing.

Abstract: Provided is an emotional-speech synthesizing device including: a sentence recognition unit that recognizes a sentence that is input; a word emotion determination unit that calculates probability vector of an emotion that is pre-defined for each word that makes up the recognized sentence and estimates the emotion and a rhythm based on the probability vector; and an emotional-speech synthesizing unit. The emotional-speech synthesizing unit calculates in stages degrees of similarity in the emotion and the rhythm between the adjacent words based on context information on the recognized sentence, applies weight to a phoneme candidate corresponding to the each word based on the degrees of the similarity and the probability vector, selects the phoneme candidate that has a minimum target pitch, minimum duration time, a minimum distance value of a target pitch contour, and thus synthesizes an emotional speech that corresponds to the recognized sentence in optimal units.

Abstract: A system and method for identifying special information is provided. Endpoints are defined within a voice recording. One or more of the endpoints are identified within the voice recording and the voice recording is partitioned into segments based on the identified endpoints. Elements of text are identified by applying speech recognition to each of the segments and a list of prompt list candidates are applied to the text elements. The segments with text elements that match one or more prompt list candidates are identified. Portions of the voice recording following the prompt list candidates that include special information are identified and the special information is rendered unintelligible within the voice recording.

Abstract: According to an in-vehicle control apparatus, it is configured to receive information of a plurality of operation keys displayed on a display, to identify from a result of voice recognition of a user's speech, the operation key corresponding thereto on the basis of the information of the operation keys, and to execute a function assigned to the operation key upon receiving an instruction for execution by the user using a decision key, so that it is possible to perform an operation without making contact directly with the plurality of operation keys displayed on the display and without the need of a complex key operation. Further, solely by mounting just one decision key on a steering wheel, it becomes possible to safely perform the operation about the operation key displayed on the display without the user releasing his/her hand from the steering wheel during driving.

Abstract: A first communication path for receiving a communication is established. The communication includes speech, which is processed. A speech pattern is identified as including a voice-command. A portion of the speech pattern is determined as including the voice-command. That portion of the speech pattern is separated from the speech pattern and compared with a second speech pattern. If the two speech patterns match or resemble each other, the portion of the speech pattern is accepted as the voice-command. An operation corresponding to the voice-command is determined and performed. The operation may perform an operation on a remote device, forward the voice-command to a remote device, or notify a user. The operation may create a second communication path that may allow a headset to join in a communication between another headset and a communication device, several headsets to communicate with each other, or a headset to communicate with several communication devices.

Abstract: A first communication path for receiving a communication is established. The communication includes speech, which is processed. A speech pattern is identified as including a voice-command. A portion of the speech pattern is determined as including the voice-command. That portion of the speech pattern is separated from the speech pattern and compared with a second speech pattern. If the two speech patterns match or resemble each other, the portion of the speech pattern is accepted as the voice-command. An operation corresponding to the voice-command is determined and performed. The operation may perform an operation on a remote device, forward the voice-command to a remote device, or notify a user. The operation may create a second communication path that may allow a headset to join in a communication between another headset and a communication device, several headsets to communicate with each other, or a headset to communicate with several communication devices.

Abstract: The subject matter of this specification can be implemented in, among other things, a computer-implemented method for correcting words in transcribed text including receiving speech audio data from a microphone. The method further includes sending the speech audio data to a transcription system. The method further includes receiving a word lattice transcribed from the speech audio data by the transcription system. The method further includes presenting one or more transcribed words from the word lattice. The method further includes receiving a user selection of at least one of the presented transcribed words. The method further includes presenting one or more alternate words from the word lattice for the selected transcribed word. The method further includes receiving a user selection of at least one of the alternate words. The method further includes replacing the selected transcribed word in the presented transcribed words with the selected alternate word.

Abstract: A method of recognizing continuous digits uttered by a speaker using an automatic speech recognition (ASR) system includes receiving continuous digits via a microphone as speech from a user; detecting that recognition of one or more of the continuous digits falls below a predetermined confidence threshold; prompting the user to identify the continuous digits using a body gesture; detecting the body gesture made by the user; and identifying one or more of the continuous digits based on the body gesture.

Abstract: Non-acoustic data from a vicinity of speech input is obtained. A subject speaker is identified as the source of the speech input from the obtained non-acoustic data by detecting mouth motion on one or more faces segmented from the non-acoustic data by comparing a first pixel intensity associated at a first time with a second pixel intensity at a second time, and selecting a face corresponding to the subject speaker from the one or more faces in response to a determination that a number of significantly changed pixels between the first pixel intensity and the second pixel intensity exceeds a threshold. A demographic is assigned to the subject speaker based on an analysis of one or more non-acoustic attributes of the subject speaker extracted from the non-acoustic data. The speech input is processed using a speech recognition system adjusted using a model selected based on the demographic.

Abstract: An approach is provided for detecting a voice call directed to a user. The approach involves presenting a user interface for interacting with the voice call, wherein the user interface includes a control option for selecting a pre-recorded word or phrase from the user; for generating a custom-created audio word or phrase from one or more phonemes pre-recorded by the user; or a combination thereof. The approach also involves interjecting the pre-recorded word or phrase, the custom-created audio word or phrase, or a combination thereof into the voice call.

Abstract: Portable recording devices used by a person, such as a law enforcement officer, may be automatically activated to capture data assets. A dispatch request may be sent from a server to a computing device used by person equipped with a portable recording device. Subsequently, the server may determine whether the portable recording device is activated by the computing device to record a data asset. In response to a determination that there is a lack of activation of the portable recording device by the computing device to record the data asset following the dispatch request, the server may activate the portable recording device of the person to record the data asset.

Abstract: The present disclosure relates to training a speech recognition system. A system that includes an automated speech recognizer and receives data from a client device. The system determines that at least a portion of the received data is likely sensitive data. Before the at least a portion of the received data is deleted, the system provides the at least a portion of the received data to a model training engine that trains recognition models for the automated speech recognizer. After the at least a portion of the received data is provided, the system deletes the at least a portion of the received data.

Abstract: The disclosed embodiments illustrate methods and systems for summary generation of a real-time conversation. The method includes receiving a real-time conversation from a plurality of computing devices over a communication network. The method further includes determining one or more first features of the real-time conversation between at least a first user and a second user. The method further includes extracting one or more second features from the one or more first features, based on one or more pre-defined criteria. The method further includes generating a summary content of the real-time conversation, based on at least the extracted one or more second features and one or more annotations associated with the determined one or more first features by use of one or more trained classifier. Further, the method includes rendering the generated summary content on a user interface displayed on at least one of the plurality of computing devices.

Abstract: A speech recognition apparatus and method. The speech recognition apparatus includes a first recognizer configured to generate a first recognition result of an audio signal, in a first linguistic recognition unit, by using an acoustic model, a second recognizer configured to generate a second recognition result of the audio signal, in a second linguistic recognition unit, by using a language model, and a combiner configured to combine the first recognition result and the second recognition result to generate a final recognition result in the second linguistic recognition unit and to reflect the final recognition result in the language model. The first linguistic recognition unit may be a same linguistic unit type as the second linguistic recognition unit. The first recognizer and the second recognizer are configured in a same neural network and simultaneously/collectively trained in the neural network using audio training data provided to the first recognizer.

Abstract: A method for improving speech recognition by a speech recognition system includes obtaining a voice sample from a speaker; storing the voice sample of the speaker as a voice model in a voice model database; identifying an area from which sound matching the voice model for the speaker is coming; providing one or more audio signals corresponding to sound received from the identified area to the speech recognition system for processing.

Abstract: In a method of diarization of audio data, audio data is segmented into a plurality of utterances. Each utterance is represented as an utterance model representative of a plurality of feature vectors. The utterance models are clustered. A plurality of speaker models are constructed from the clustered utterance models. A hidden Markov model is constructed of the plurality of speaker models. A sequence of identified speaker models is decoded.

Abstract: A method of controlling a terminal is provided. The method includes analyzing a sensed voice when a voice is sensed, recognizing a context based a result of the analysis, and performing a predetermined control operation based on the recognized context.

Abstract: An apparatus for detecting a sound in an acoustical environment includes a microphone array configured to detect an audio signal in the acoustical environment. The apparatus also includes a processor configured to determine an angular location of a sound source of the audio signal. The angular location is relative to the microphone array. The processor is also configured to determine at least one reverberation characteristic of the audio signal. The processor is further configured to determine a distance, relative to the microphone array, of the sound source along an axis associated with the angular location based on the at least one reverberation characteristic.

Abstract: Provided are, among other things, systems, methods and techniques for compressing an audio signal. According to one representative embodiment, an audio signal that includes quantization indexes, identification of segments of such quantization indexes, and indexes of entropy codebooks that have been assigned to such segments is obtained, with a single entropy codebook index having been assigned to each such segment. Potential merging operations in which adjacent ones of the segments potentially would be merged with each are identified, and bit penalties for the potential merging operations are estimated. At least one of the potential merging operations is performed based on the estimated bit penalties, thereby obtaining a smaller updated set of segments of quantization indexes and corresponding assigned codebooks. The quantization indexes in each of the segments in the smaller updated set are then entropy encoded by using the corresponding assigned entropy codebooks, thereby compressing the audio signal.

Abstract: The present document relates to audio signal processing in general, and to the concealment of artifacts that results from loss of audio packets during audio transmission over a packet-switched network, in particular. A method (200) for concealing one or more consecutive lost packets (412, 413) is described. A lost packet (412) is a packet which is deemed to be lost by a transform-based audio decoder. Each of the one or more lost packets (412, 413) comprises a set of transform coefficients (313). A set of transform coefficients (313) is used by the transform-based audio decoder to generate a corresponding frame (412, 413) of a time domain audio signal. The method (200) comprises determining (205) for a current lost packet (412) of the one or more lost packets (412, 413) a number of preceding lost packets from the one or more lost packets (313); wherein the determined number is referred to as a loss position.

Abstract: Methods and audio processing units for generating an object based audio program including conditional rendering metadata corresponding to at least one object channel of the program, where the conditional rendering metadata is indicative of at least one rendering constraint, based on playback speaker array configuration, which applies to each corresponding object channel, and methods for rendering audio content determined by such a program, including by rendering content of at least one audio channel of the program in a manner compliant with each applicable rendering constraint in response to at least some of the conditional rendering metadata. Rendering of a selected mix of content of the program may provide an immersive experience.

Abstract: A digital watermark embedding device includes a synthesized voice generating unit that outputs a synthesized voice according to an input text and outputs phoneme-based alignment regarding phonemes included in the synthesized voice; an estimating unit that estimates whether or not a potentially risky expression is included in the input text, and outputs a potentially risky segment in which the potentially risky expression is estimated to be included; an embedding control unit that associates the potentially risky segment with the phoneme-based alignment, and decides and outputs an embedding time for embedding a watermark in the synthesized voice; and an embedding unit that embeds a digital watermark in the synthesized voice at a time specified as the embedding time for the synthesized voice.

Abstract: A method is provided. The method includes obtaining a low-band spectrum of an audio signal in which a low-band signal is frequency transformed; obtaining phase information of a high-band spectrum of the audio signal based on the low-band spectrum; and outputting a bitstream that comprises the phase information of the high-band spectrum.

Abstract: In this invention, the design of the Huffman table can be done offline with a large input sequence database. The range of the quantization indices (or differential indices) for Huffman coding is identified. For each value of range, all the input signal which have the same range will be gathered and the probability distribution of each value of the quantization indices (or differential indices) within the range is calculated. For each value of range, one Huffman table is designed according to the probability. And in order to improve the bits efficiency of the Huffman coding, apparatus and methods to reduce the range of the quantization indices (or differential indices) are also introduced.

Abstract: A vector joint encoding/decoding method and a vector joint encoder/decoder are provided, more than two vectors are jointly encoded, and an encoding index of at least one vector is split and then combined between different vectors, so that encoding idle spaces of different vectors can be recombined, thereby facilitating saving of encoding bits, and because an encoding index of a vector is split and then shorter split indexes are recombined, thereby facilitating reduction of requirements for the bit width of operating parts in encoding/decoding calculation.

Abstract: An audio signal transmission device for encoding an audio signal includes an audio encoding unit that encodes an audio signal and a side information encoding unit that calculates and encodes side information from a look-ahead signal. An audio signal receiving device for decoding an audio code and outputting an audio signal includes: an audio code buffer that detects packet loss based on a received state of an audio packet, an audio parameter decoding unit that decodes an audio code when an audio packet is correctly received, a side information decoding unit that decodes a side information code when an audio packet is correctly received, a side information accumulation unit that accumulates side information obtained by decoding a side information code, an audio parameter missing processing unit that outputs an audio parameter upon detection of audio packet loss, and an audio synthesis unit that synthesizes decoded audio from the audio parameter.

Abstract: In one example, a method includes obtaining an audio signal comprising a plurality of elements; generating a first Higher-Order Ambisonics (HOA) soundfield that represents the audio signal; selecting a set of elements of the audio signal for encoding in a non-Higher-Order Ambisonics (HOA) domain; generating, based on the selected set of elements and a set of spatial positioning vectors, a second HOA soundfield that represents the selected set of elements; generating a third HOA soundfield that represents a difference between the first HOA soundfield and the second HOA soundfield; and generate a coded audio bitstream that includes a representation of the selected set of elements in the non-HOA domain, an indication of the set of spatial positioning vectors, and a representation of the third HOA soundfield.