Abstract: Certain example embodiments relate to an acoustic wall assembly that uses active and/or passive sound reverberation to achieve noise-disruptive functionality, and/or a method of making and/or using the same. With the active approach, sound waves in a given frequency range are detected by a sound masking circuit. Responsive to detection of such sound waves, an air pump (e.g., speaker) is used to pump air in the wall assembly to actively mask the detected sound waves via reverberation and/or the like. The wall assembly may include one, two, or more walls, and the walls may be partial or full walls. With the passive approach, sound waves in a given frequency range are disrupted via features (e.g., holes, slits, etc.) formed in and/or on a wall itself. These techniques may be used together or separately, in different example embodiments.

Abstract: A noise and vibration sensor arrangement which is configured to operate with a road noise control system, and includes four acceleration sensors each configured to generate at least one output signal representative of at least one of accelerations, motions and vibrations that act on the respective acceleration sensor. The arrangement includes a vehicle subframe structure having a shape that is axisymmetric to a first axis and that has a maximum extent along a second axis. The first and the second axis are perpendicular. The four sensors are attached to the subframe structure at positions that correspond to four corners of a virtual rectangle. The virtual rectangle has two perpendicular centerlines, one of the centerlines being in line with the first axis of the subframe structure. The extent of the virtual rectangle along the other centerline is less than fifty percent of the subframe structure's maximum extent along the second axis.

Abstract: An example active road noise control includes generating with a sensor arrangement a primary sense signal representative of at least one of accelerations, motions and vibrations that occur at a first position, and providing a noise reducing signal by processing the primary sense signal according to an adaptive mode of operation or a non-adaptive mode of operation. It further includes generating within the vehicle body noise reducing sound at the second position from the noise reducing signal, evaluating the primary sense signal and controlling the processing of the primary sense signal so that the primary sense signal is processed in the adaptive mode of operation when the magnitude of the primary sense signal undercuts a first threshold and in the non-adaptive mode of operation when the magnitude of the primary sense signal exceeds a second threshold, the first threshold being equal to or smaller than the second threshold.

Abstract: A vehicle sound effect generation apparatus includes a running state detecting unit that detects a running state of a vehicle, an adjustment wave sound selector that selects one or more adjustment wave sounds having a half-order frequency component or an integer-order frequency component, a sound effect generation unit that synthesizes a fundamental wave sound having a fundamental frequency component with the one or more adjustment wave sounds selected by the adjustment wave sound selector, and an inhibition condition determining unit that can determine a condition that correspondence between an amount of an operation performed by a vehicle occupant and a behavior of the vehicle is inhibited. When the inhibition condition determining unit determines the condition that the correspondence between the amount of the operation and the behavior of the vehicle is inhibited, the adjustment wave sound selector reduces the output characteristics of the adjustment wave sounds selected.

Abstract: Systems, methods, and computer-readable storage media for intelligent caching of concatenative speech units for use in speech synthesis. A system configured to practice the method can identify, in a local cache of text-to-speech units for a text-to-speech voice an absent text-to-speech unit which is not in the local cache. The system can request from a server the absent text-to-speech unit. The system can then synthesize speech using the text-to-speech units and a received text-to-speech unit from the server.

Abstract: A method and apparatus are described for detecting voice related vibration in the upper region of the chest and synthesizing human speech. The innovation finds its use in speech rehabilitation applications among others, specifically in speech impairments and speech disability arising due to accident, congenital defects or other reasons. A set of piezoelectric based sensors are placed on an upper region of the chest atop or near sound tendons. The sensors pick up the vibrations in the sound tendons and convert the vibrations into electrical output signals. These signals are filtered, amplified and processed using the signal recognition unit. Subsequently, a set of parameters are extracted and used to generate speech or a written text. The sensors incorporate piezoelectric or other transducing materials. These sensors are externally affixed to a human body surface corresponding to the position of the sounds tendons in the upper chest/neck region.

Abstract: Systems and methods are provided for associating a phonetic pronunciation with a name by receiving the name, mapping the name to a plurality of monosyllabic components that are combinable to construct the phonetic pronunciation of the name, receiving a user input to select one or more of the plurality, and combining the selected one or more of the plurality of monosyllabic components to construct the phonetic pronunciation of the name.

Abstract: Systems and methods for identifying content corresponding to a language are provided. Language spoken by a first user based on verbal input received from the first user is automatically determined with voice recognition circuitry. A database of content sources is cross-referenced to identify a content source associated with a language field value that corresponds to the determined language spoken by the first user. The language field in the database identifies the language that the associated content source transmits content to a plurality of users. A representation of the identified content source is generated for display to the first user.

Abstract: An image display apparatus, a method for driving the same, and a computer readable recording medium are provided. The image display apparatus includes a speech acquirer configured to acquire a speech command created by a user, a speech recognition executor configured to acquire text information which has a phonetic symbol that is the same as or similar to a phonetic symbol of a text-based recognition result corresponding to the acquired speech command and is expressed in a form that is different from a form of the text-based recognition result, and an operation performer configured to perform an operation corresponding to the acquired text information.

Abstract: An automatic speech recognition (ASR) system may add new words to an ASR system by identifying words with similar usage and replicating the variations of the identified words to create new words. A new word that is used similarly to a known word may be varied to create new word forms that are similar to the word forms of a known word. The new word forms may then be incorporated into an ASR model to allow the ASR system to recognize those words when they are detected in speech. Such a system may allow flexible incorporation and recognition of varied forms of new words entering a general lexicon.

Abstract: A system is provided that trains a spoken language understanding (SLU) classifier. A corpus of user utterances is received. For each of the user utterances in the corpus, the user utterance is semantically parsed, and the result of this semantic parsing is represented as a rooted semantic parse graph. The parse graphs representing all of the user utterances in the corpus are then combined into a single corpus graph that represents the semantic parses of the entire corpus. The user utterances in the corpus are then clustered into intent-wise homogeneous groups of user utterances, where this clustering includes finding subgraphs in the corpus graph that represent different groups of user utterances, and each of these different groups has a similar user intent. The intent-wise homogeneous groups of user utterances are then used to train the SLU classifier, and the trained SLU classifier is output.

Abstract: An electronic device includes a memory configured to store a user pronunciation lexicon, a voice input unit configured to receive a user's uttered voice, and a processor configured to extract a user pronunciation pattern from the received uttered voice and to update the user pronunciation lexicon according to a pronunciation pattern rule generated based on the extracted pronunciation pattern.

Abstract: System and method to adjust an automatic speech recognition (ASR) engine, the method including: receiving social network information from a social network; data mining the social network information to extract one or more characteristics; inferring a trend from the extracted one or more characteristics; and adjusting the ASR engine based upon the inferred trend. Embodiments of the method may further include: receiving a speech signal from a user; and recognizing the speech signal by use of the adjusted ASR engine. Further embodiments of the method may include: producing a list of candidate matching words; and ranking the list of candidate matching words by use of the inferred trend.

Abstract: A terminal comprises: a speech receiving unit that receives speech in a locked state; a voiceprint authentication unit that performs voiceprint authentication based on the speech received in the locked state and determining whether or not a user is legitimate; a speech recognition unit that performs speech recognition of the speech received in the locked state; and an execution unit that executes an application using a result of the speech recognition.

Abstract: Methods, systems, and apparatus, including computer programs encoded on computer storage media, for generating representation of acoustic sequences. One of the methods includes: receiving an acoustic sequence, the acoustic sequence comprising a respective acoustic feature representation at each of a plurality of time steps; processing the acoustic feature representation at an initial time step using an acoustic modeling neural network; for each subsequent time step of the plurality of time steps: receiving an output generated by the acoustic modeling neural network for a preceding time step, generating a modified input from the output generated by the acoustic modeling neural network for the preceding time step and the acoustic representation for the time step, and processing the modified input using the acoustic modeling neural network to generate an output for the time step; and generating a phoneme representation for the utterance from the outputs for each of the time steps.

Abstract: Methods, systems, and apparatus, including computer programs encoded on a computer storage medium, relating to generating log-linear models. In some implementations, n-gram parameter values derived from an n-gram language model are obtained. N-gram features for a log-linear language model are determined based on the n-grams corresponding to the obtained n-gram parameter values. A weight for each of the determined n-gram features is determined, where the weight is determined based on (i) an n-gram parameter value that is derived from the n-gram language model and that corresponds to a particular n-gram, and (ii) an n-gram parameter value that is derived from the n-gram language model and that corresponds to an n-gram that is a sub-sequence within the particular n-gram. A log-linear language model having the determined n-gram features is generated, where the determined n-gram features in the log-linear language model have weights that are initialized based on the determined weights.

Abstract: Techniques for providing virtual assistants to assist users during a voice communication between the users. For instance, a first user operating a device may establish a voice communication with respective devices of one or more additional users, such as with a device of a second user. For instance, the first user may utilize her device to place a telephone call to the device of the second user. A virtual assistant may also join the call and, upon invocation by a user on the call, may identify voice commands from the call and may perform corresponding tasks for the users in response.

Abstract: In some implementations, a method is performed by computing devices, and includes receiving a first message indicating that an improper voice command has been detected by a first device, identifying a user account associated with the first device based on the first message, identifying a second device associated with the user account, and, in response to receiving the first message indicating that the voice command detected by the first device is improper, sending a second message to the second device, the second message indicating that the voice command should not be performed.

Abstract: Methods, apparatus, and computer-readable media (transitory and non-transitory) are provided herein for reducing latency caused by switching input modalities. In various implementations, a first input such as text input may be received at a first modality of a multimodal interface provided by an electronic device. In response to determination that the first input satisfies one or more criteria, the electronic device may preemptively establish a session between the electronic device and a query processor configured to process input received at a second modality (e.g., voice input) of the multimodal interface. In various implementations, the electronic device may receive a second input (e.g., voice input) at the second modality of the multimodal interface, initiate processing of at least a portion of the second input at the query processor within the session, and build a complete query based on output from the query processor.

Abstract: Methods, systems, and apparatus, including computer programs encoded on a computer storage medium, for hotword detection on multiple devices are disclosed. In one aspect, a method includes the actions of receiving, by a first computing device, audio data that corresponds to an utterance. The actions further include determining a first value corresponding to a likelihood that the utterance includes a hotword. The actions further include receiving a second value corresponding to a likelihood that the utterance includes the hotword, the second value being determined by a second computing device. The actions further include comparing the first value and the second value. The actions further include based on comparing the first value to the second value, initiating speech recognition processing on the audio data.

Abstract: Disclosed herein are example techniques to provide contextual information corresponding to a voice command. An example implementation may involve receiving voice data indicating a voice command, receiving contextual information indicating a characteristic of the voice command, and determining a device operation corresponding to the voice command. Determining the device operation corresponding to the voice command may include identifying, among multiple zones of a media playback system, a zone that corresponds to the characteristic of the voice command, and determining that the voice command corresponds to one or more particular devices that are associated with the identified zone. The example implementation may further involve causing the one or more particular devices to perform the device operation.

Abstract: Systems and method of diarization of audio files use an acoustic voiceprint model. A plurality of audio files are analyzed to arrive at an acoustic voiceprint model associated to an identified speaker. Metadata associate with an audio file is used to select an acoustic voiceprint model. The selected acoustic voiceprint model is applied in a diarization to identify audio data of the identified speaker.

Abstract: Systems and methods of diarization using linguistic labeling include receiving a set of diarized textual transcripts. A least one heuristic is automatedly applied to the diarized textual transcripts to select transcripts likely to be associated with an identified group of speakers. The selected transcripts are analyzed to create at least one linguistic model. The linguistic model is applied to transcripted audio data to label a portion of the transcripted audio data as having been spoken by the identified group of speakers. Still further embodiments of diarization using linguistic labeling may serve to label agent speech and customer speech in a recorded and transcripted customer service interaction.

Abstract: The present disclosure provide a signal processing method and apparatus. The method includes: determining a total quantity of to-be-allocated bits corresponding to a current frame; implementing primary bit allocation on to-be-processed sub-bands; performing a primary information unit quantity determining operation for each sub-band that has undergone the primary bit allocation; selecting sub-bands for secondary bit allocation from the to-be-processed sub-bands according to at least one of a sub-band characteristic of each sub-band of the to-be-processed sub-bands or the total quantity of surplus bits; implementing secondary bit allocation on the sub-bands for secondary bit allocation; and performing a secondary information unit quantity determining operation for each sub-band of the sub-bands for secondary bit allocation.

Abstract: In general, techniques are described for crossfading sets of spherical harmonic coefficients. An audio encoding device or audio decoding device comprising a memory and a processor may be configured to perform the techniques. The memory may be configured to store a first set of spherical harmonic coefficients (SHCs) and a second set of SHCs. The first set of SHCs describe a first sound field. The second set of SHCs describe a second sound field. The processor may be configured to crossfade between the first set of SHCs and a second set of SHCs to obtain a first set of crossfaded SHCs.

Abstract: An apparatus for generating a decoded two-channel signal includes: an audio processor for decoding an encoded two-channel signal to obtain a first set of first spectral portions; a parametric decoder for providing parametric data for a second set of second spectral portions and a two-channel identification identifying either a first or a second different two-channel representation for the second spectral portions; and a frequency regenerator for regenerating a second spectral portion depending on a first spectral portion of the first set of first spectral portions, the parametric data for the second portion and the two-channel identification for the second portion.

Abstract: Soundfield signals such as e.g. Ambisonics carry a representation of a desired sound field. The Ambisonics format is based on spherical harmonic decomposition of the soundfield, and Higher Order Ambisonics (HOA) uses spherical harmonics of at least 2nd order. However, commonly used loudspeaker setups are irregular and lead to problems in decoder design. A method for improved decoding an audio soundfield representation for audio playback comprises calculating a panning function (W) using a geometrical method based on the positions of a plurality of loudspeakers and a plurality of source directions, calculating a mode matrix (?) from the loudspeaker positions, calculating a pseudo-inverse mode matrix (?+) and decoding the audio soundfield representation. The decoding is based on a decode matrix (D) that is obtained from the panning function (W) and the pseudo-inverse mode matrix (?+).

Abstract: Present disclosure provide a linear prediction-based noise signal processing method includes: acquiring a noise signal, and obtaining a linear prediction coefficient according to the noise signal; filtering the noise signal according to the linear prediction coefficient, to obtain a linear prediction residual signal; obtaining a spectral envelope of the linear prediction residual signal according to the linear prediction residual signal; and encoding the spectral envelope of the linear prediction residual signal. According to the noise processing method, the noise generation method, the encoder, the decoder, and the encoding and decoding system that are in the embodiments of the present disclosure, more spectral details of an original background noise signal can be recovered, so that comfort noise can be closer to original background noise in terms of subjective auditory perception of a user, and subjective perception quality of the user is improved.

Abstract: To provide a transmission method of an arbitrary signal using an acoustic sound hardly affecting the mood (quality) of the original acoustic sound even if an arbitrary signal within an audible frequency range is combined into the acoustic sound such as music. The method comprises a step of finding a separable sound, of which a fundamental element (fundamental sound) b2 contributing mainly to recognition of a single sound and an additional element contributing collaterally to recognition of the single sound are separable on temporal axis or frequency axis, among plural sounds composing the acoustic sound, or inserting a separable sound into the plural sounds composing the acoustic sound. The additional element of the separable sound found or inserted is transcribed by a signal pattern b1-1, b1-2 of the arbitrary signal. By means of the acoustic sound of which additional element is transcribed, the arbitrary signal b1-1, b1-2 is transmitted.

Abstract: Methods and apparatus to audio watermarking and watermark detection and extracted are described herein. According to an example method, an identifier is encoded in media content when a different identifier has been previously encoded. According to another example method, messages decoded from media content are validated to provide improved decoding accuracy. In another example method, decoded symbols are stored in memory and synchronization symbols are located to detect a message encoded in media content.

Abstract: In one aspect, the invention divides an audio signal into auditory events, each of which tends to be perceived as separate and distinct, by calculating the spectral content of successive time blocks of the audio signal, calculating the difference in spectral content between successive time blocks of the audio signal, and identifying an auditory event boundary as the boundary between successive time blocks when the difference in the spectral content between such successive time blocks exceeds a threshold. In another aspect, the invention generates a reduced-information representation of an audio signal by dividing an audio signal into auditory events, each of which tends to be perceived as separate and distinct, and formatting and storing information relating to the auditory events. Optionally, the invention may also assign a characteristic to one or more of the auditory events. Auditory events may be determined according to the first aspect of the invention or by another method.

Abstract: A coding apparatus, including a memory and a processor that, when executing instructions stored in the memory, performs operations including encoding low-band transform coefficients in a first band and calculating, for each extension-band subband obtained by splitting an extension band, a threshold amplitude based on an analysis of statistics on extension-band transform coefficients included in the subband.

Abstract: A multiresolution coding and modulation system for preserving audio continuity in a digital wireless audio system in harsh RF environments is provided that includes an audio codec that produces codewords having bits differing in perceptual importance coupled with a non-uniform hierarchical modulation scheme. When the wireless audio system is operating in a harsh RF environment, the RF SNR for decoding bits having a high perceptual importance may be decreased while the RF SNR for decoding bits having a low perceptual importance may be increased, while having no adverse effect on the latency of the wireless audio system. The audio may be subjectively degraded instead of muting the audio, which mitigates intermittent interference and multipath fading issues.

Abstract: Embodiments relate to an audio processing unit that includes a buffer, bitstream payload deformatter, and a decoding subsystem. The buffer stores at least one block of an encoded audio bitstream. The block includes a fill element that begins with an identifier followed by fill data. The fill data includes at least one flag identifying whether enhanced spectral band replication (eSBR) processing is to be performed on audio content of the block. A corresponding method for decoding an encoded audio bitstream is also provided.

Abstract: A videoconference apparatus at a first location detects audio from a location and determines whether the sound should be included in an audio-video stream sent to a second location, or excluded as an interfering noise. Determining whether to include the audio involves using a face detector to see if there is a face at the source of the sound. If a face is present, the audio data from the location will be transmitted to the second location. If a face is not present, additional motion checks are performed to determine whether the sound corresponds to a person talking, (such as a presenter at a meeting), or whether the sound is instead unwanted noise.

Abstract: Method and apparatus are disclosed for determining and accounting for distortions to an audio signal due to the geometric properties of the interior cabin of a vehicle. An example vehicle includes a microphone, a seat having a plurality of seat positions, and a processor. The processor is configured to determine a first seat position corresponding to a point in time at which an audio signal is received, determine a cabin impulse response corresponding to the first seat position, and determine a filtered audio signal based on the cabin impulse response and the audio signal.

Abstract: The privacy-preserving energy-efficient speaker implementations described herein improve user privacy while a user is listening to audio and can reduce the energy necessary to output the audio. This can be done by using parametric speakers and/or traditional loud-speakers. Signal splitting and masking can be used to improve user privacy. Additionally, a signal modulation technique which significantly reduces power requirements to output an audio signal, especially in the context of using parametric speakers, can also be employed.

Abstract: The disclosure provides a method and an apparatus for detecting a voice activity in an input audio signal composed of frames. A noise characteristic of the input signal is determined based on a received frame of the input audio signal. A voice activity detection (VAD) parameter is derived based on the noise characteristic of the input audio signal using an adaptive function. The derived VAD parameter is compared with a threshold value to provide a voice activity detection decision. The input audio signal is processed according to the voice activity detection decision.

Abstract: The present technique relates to a frequency band extension apparatus, a frequency band extension method, and a program which are configured to more easily obtain a high quality sound signal. An input signal may be divided into sub-band signals of a plurality of sub-bands, powers of high frequency sub-bands of the input signal may be estimated based on feature values extracted from the input signal to obtain high frequency sub-band power estimation values, the high frequency sub-band powers obtained from the sub-band signals of high-frequency sub-bands of the input signal may be compared with the high frequency sub-band power estimation values, and a high-frequency signal of the input signal may be generated based on a result of the comparison and the sub-band signals.

Abstract: An autocorrelation calculating part calculates autocorrelation Ro(i) from an input signal. A predictive coefficient calculating part performs linear predictive analysis using modified autocorrelation R?o(i) obtained by multiplying the autocorrelation Ro(i) by a coefficient wo(i). Here, it is assumed that a case where, for at least part of each order i, the coefficient wo(i) corresponding to each order i monotonically increases as a value having negative correlation with a fundamental frequency of an input signal in a current frame or a past frame increases and a case where the coefficient wo(i) monotonically decreases as a value having positive correlation with a pitch gain in a current frame or a past frame increases, are included.

Abstract: An autocorrelation calculating part calculates autocorrelation Ro(i) from an input signal. A predictive coefficient calculating part performs linear predictive analysis using modified autocorrelation R?o(i) obtained by multiplying the autocorrelation Ro(i) by a coefficient wo(i). Here, it is assumed that a case where, for at least part of each order i, the coefficient wo(i) corresponding to each order i monotonically increases as a value having negative correlation with a fundamental frequency of an input signal in a current frame or a past frame increases and a case where the coefficient wo(i) monotonically decreases as a value having positive correlation with a pitch gain in a current frame or a past frame increases, are included.

Abstract: A neural network model, such as a deep neural network (DNN), is trained using many speech examples to perform beam selection in a microphone array-based speech processing system. The DNN is trained using many different speech examples that are labeled with position or direction information relative to a training microphone array. The DNN may then be trained to recognize a direction of incoming speech so that at runtime the trained DNN may process input audio data from a microphone array and may output to a beam selector an indicator of the desired beam that may be selected for further processing. The DNN may be configured to output a beam index and/or coordinates (or other position data) corresponding to an estimated location of the detected speech. The DNN may also be configured to output acoustic unit data corresponding to speech units (for example corresponding to phonemes, senons, etc. such as those of a detected wakeword or other word).

Abstract: A method of determining, by an electronic device, an audio event is disclosed. The method may include receiving an input sound from a sound source by a plurality of sound sensors. The method may also extracting, by a processor, at least one sound feature from the received input sound, determining, by the processor, location information of the sound source based on the input sound received by the sound sensors, determining, by the processor, the audio event indicative of the input sound based on the at least one sound feature and the location information, and transmitting, by a communication unit, a notification of the audio event to an external electronic device.

Abstract: System(s) and method(s) for classifying noise data of human crowd are disclosed. Noise data is captured from one or more sources and features are extracted by using computation techniques. The features comprise spectral domain features and time domain features. Classification models are developed by using each of the spectral domain features and the time domain features. Discriminative information with respect to the noise data is extracted by using the classification models. A performance matrix is computed for each of the classification model. The performance matrix comprises classified noise elements with respect to the noise data. Each classified noise element is associated with a classification performance score with respect to a spectral domain feature, a time domain feature, and fusion of features and scores. The classified noise elements provide the classification of the noise data.

Abstract: This disclosure generally relates to a portable device. Specifically, this disclosure generally relates to a portable word counter device. The portable word counter device includes a digital microcontroller circuit. The digital microcontroller circuit includes a syllable detector detecting syllables in spoken speech. The syllable detector aggregates a number of detected syllables and applies a syllable to word counted ratio. Based on the syllable to word counted ratio, the syllable detector determines a number of words spoken, and transmits the number of words spoken to a mobile device.

Abstract: A system for determining an endpoint of an utterance during automatic speech recognition (ASR) processing that accounts for the direction and duration of the incoming speech. Beamformers of the ASR system may identify a source direction of the audio. The system may track the duration speech has been received from that source direction so that if speech is detected in another direction, the original source speech may be weighted differently for purposes of determining an endpoint of the utterance. Speech from a new direction may be discarded or treated like non-speech for purposes of determining an endpoint of speech from an original direction.

Abstract: The tone arm holder includes a saddle with a hook portion that defines an open end. A resilient dividing member is located on the top surface of the saddle. The space between the resilient dividing member and the hook define a lock seat configured and arranged for frictional engagement, receipt and storage of a tone arm therein. The opposing side of the saddle from the resilient member defines a performance seat configured and arranged for freely supporting a tone arm whereby a tone arm may be placed on and removed from the performance seat without frictional engagement of the tone arm to the holder. The tone arm is freely movable between the performance seat at a first position and the lock seat at a second position by traversing across the resilient dividing member by exceeding the resilient force of the resilient dividing member.

Abstract: Mounting a data storage medium having information recorded thereon, where: the information is formatted according to a self-describing format standard that includes first write and first read functions, the information includes first and second datasets, the first dataset includes a first file mark, an index, and an empty space between the second dataset and a combination of the first file mark and the index; receiving, from a first application, a data block and a corresponding write command for writing the data block onto the data storage medium; and in response to receiving the write command: determining that the empty space is present in the first dataset, writing, by a second write function, the data block into the empty space, and writing a second file mark in the second dataset; wherein the empty space of the data storage medium is inaccessible to the first write function of the self-describing format standard.

Abstract: A data writer may be constructed and operated as part of a data storage device. The data writer can be positioned proximal a data storage medium. The data writer may have a write pole positioned adjacent a writer coil with the writer coil having a plurality of turns. A controller that is connected to each turn can be adapted to selectively activate less than all the coil turns in response to the data writer being positioned over a first portion of a data storage medium and selectively activate all of the coil turns in response to the data writer being positioned over a second portion of the data storage medium.

Abstract: A method of making a magnetic head, according to one embodiment, includes supplying a magnetic recording head having a gap disposed between a rowbar substrate and a closure, said rowbar substrate and closure having a tape support surface, recessing the gap below the level of the tape support surface, sputter cleaning the head, and depositing an insulation layer on the tape support surface and the recessed gap. In addition, about an entire upper surface of the insulation layer above the gap is recessed from the tape support surface.