Abstract: A headset includes a microphone that detects an acoustic signal, and converts the acoustic signal into a microphone signal, an audio processor that receives the microphone signal, and a twisted pair conductor element coupling the microphone and the audio processor. The twisted pair conductor element self-cancels a radio frequency (RF) field to prevent the RF field from entering the microphone.

Abstract: A mobile terminal apparatus portable by a user includes a recording unit, a memory, an estimating unit, and an output unit. The recording unit records a sound that has been produced by another apparatus. The memory stores a cause of an unusual noise that occurs in the other apparatus and intensities of individual frequency components of the unusual noise in association with each other. The estimating unit estimates a cause of the sound that has been produced by the other apparatus, on the basis of intensities of individual frequency components of the sound recorded by the recording unit and content stored in the memory, the intensities being calculated from the sound. The output unit outputs information about the cause of the sound estimated by the estimating unit.

Abstract: A waveguide mounted in the area beneath the helm of a boat directs acoustic output from one or more loudspeakers to its windshield where the sound can be contained, reinforced and directed toward the entertainment areas of the boat.

Abstract: Provided are methods and systems for generating Direct-to-Reverberant Ratio (DRR) estimates. The methods and systems use a null-steered beamformer to produce accurate DRR estimates across a variety of room sizes, reverberation times, and source-receiver distances. The DRR estimation algorithm uses spatial selectivity to separate direct and reverberant energy and account for noise separately. The formulation considers the response of the beamformer to reverberant sound and the effect of noise. The DRR estimation algorithm is more robust to background noise than existing approaches, and is applicable where a signal is recorded with two or more microphones, such as with mobile communications devices, laptop computers, and the like.

Abstract: Disclosed herein are systems, methods, and non-transitory computer-readable storage media for reducing latency in web-browsing TTS systems without the use of a plug-in or Flash® module. A system configured according to the disclosed methods allows the browser to send prosodically meaningful sections of text to a web server. A TTS server then converts intonational phrases of the text into audio and responds to the browser with the audio file. The system saves the audio file in a cache, with the file indexed by a unique identifier. As the system continues converting text into speech, when identical text appears the system uses the cached audio corresponding to the identical text without the need for re-synthesis via the TTS server.

Abstract: In some implementations, a language proficiency of a user of a client device is determined by one or more computers. The one or more computers then determines a text segment for output by a text-to-speech module based on the determined language proficiency of the user. After determining the text segment for output, the one or more computers generates audio data including a synthesized utterance of the text segment. The audio data including the synthesized utterance of the text segment is then provided to the client device for output.

Abstract: The disclosed embodiments relate to a method of keyword recognition in a speech signal. The method includes determining a first likelihood score and a second likelihood score of one or more features of a frame of said speech signal being associated with one or more states in a first model and one or more states in a second model, respectively. The one or more states in the first model corresponds to one or more tied triphone states and the one or more states in the second model corresponds to one or more monophone states of a keyword to be recognized in the speech signal. The method further includes determining a third likelihood score based on the first likelihood score and the second likelihood score. The first likelihood score and the third likelihood score are utilizable to determine presence of the keyword in the speech signal.

Abstract: One embodiment provides a method for generating a process learning graph and a document output from a recorded process for training a cognitive agent, the method comprising: utilizing at least one processor to execute computer code that performs the steps of: obtaining a recording of a process, wherein the recording comprises a demonstration of executing the process; generating, using the recording, the process learning graph, wherein the process learning graph identifies a process flow; generating, using the recording, the document output, wherein the document output comprises process screen transitions and process steps; and providing the process learning graph and the document output to the cognitive agent. Other aspects are described and claimed.

Abstract: Methods, systems, and apparatus, including computer programs encoded on computer storage media for speech recognition. One method includes obtaining an input acoustic sequence, the input acoustic sequence representing an utterance, and the input acoustic sequence comprising a respective acoustic feature representation at each of a first number of time steps; processing the input acoustic sequence using a first neural network to convert the input acoustic sequence into an alternative representation for the input acoustic sequence; processing the alternative representation for the input acoustic sequence using an attention-based Recurrent Neural Network (RNN) to generate, for each position in an output sequence order, a set of substring scores that includes a respective substring score for each substring in a set of substrings; and generating a sequence of substrings that represent a transcription of the utterance.

Abstract: A method for using speech disfluencies detected in speech input to assist in interpreting the input is provided. The method includes providing access to a set of content items with metadata describing the content items, and receiving a speech input intended to identify a desired content item. The method further includes detecting a speech disfluency in the speech input and determining a measure of confidence of a user in a portion of the speech input following the speech disfluency. If the confidence measure is lower than a threshold value, the method includes determining an alternative query input based on replacing the portion of the speech input following the speech disfluency with another word or phrase. The method further includes selecting content items based on comparing the speech input, the alternative query input (when the confidence measure is low), and the metadata associated with the content items.

Abstract: This disclosure describes, in part, techniques and devices for identifying recurring environmental sounds in an environment such that these sounds may be canceled out of corresponding audio signals to increase signal-to-noise ratios (SNRs) of the signals and, hence, improve automatic speech recognition (ASR) on the signals. Recurring environmental sounds may include the ringing of a mobile phone, the beeping sound of a microphone, the buzzing of a washing machine, or the like.

Abstract: Systems and methods for multi-sourced noise suppression are provided. An example system may receive streams of audio data including a voice signal and noise, the voice signal including a spoken word. The streams of audio data are provided by distributed audio devices. The system can assign weights to the audio streams based at least partially on quality of the audio streams. The weights of audio streams can be determined based on signal-to-noise ratios (SNRs). The system may further process, based on the weights, the audio stream to generate cleaned speech. Each audio device comprises microphone(s) and can be associated with the Internet of Things (IoT), such that the audio devices are Internet of Things devices. The processing can include noise suppression and reduction and echo cancellation. The cleaned speech can be provided to a remote device for further processing which may include Automatic Speech Recognition (ASR).

Abstract: A feature compensation apparatus includes a feature extractor configured to extract corrupt speech features from a corrupt speech signal with additive noise that consists of two or more frames; a noise estimator configured to estimate noise features based on the extracted corrupt speech features and compensated speech features; a probability calculator configured to calculate a correlation between adjacent frames of the corrupt speech signal; and a speech feature compensator configured to generate compensated speech features by eliminating noise features of the extracted corrupt speech features while taking into consideration the correlation between adjacent frames of the corrupt speech signal and the estimated noise features, and to transmit the generated compensated speech features to the noise estimator.

Abstract: A communication interface apparatus for a system and a plurality of users is provided. The communication interface apparatus for the system and the plurality of users includes a first process unit configured to receive voice information and face information from at least one user, and determine whether the received voice information is voice information of at least one registered user based on user models corresponding to the respective received voice information and face information; a second process unit configured to receive the face information, and determine whether the at least one user's attention is on the system based on the received face information; and a third process unit configured to receive the voice information, analyze the received voice information, and determine whether the received voice information is substantially meaningful to the system based on a dialog model that represents conversation flow on a situation basis.

Abstract: A system and method are provided for performing speech processing. A system includes an audio detection system configured to receive a signal including speech and a memory having stored therein a database of keyword models forming an ensemble of filters associated with each keyword in the database. A processor is configured to receive the signal including speech from the audio detection system, decompose the signal including speech into a sparse set of phonetic impulses, and access the database of keywords and convolve the sparse set of phonetic impulses with the ensemble of filters. The processor is further configured to identify keywords within the signal including speech based a result of the convolution and control operation the electronic system based on the keywords identified.

Abstract: A speech recognition apparatus includes a voice collection unit configured to collect voices, and a controller configured to control recognizing a speech based on signals of the voices collected during a predetermined time period from a voice collection start time, identifying whether a signal is received during the predetermined time period when an operation command corresponding to the recognized speech is not identified, determining that an early utterance occurs upon determination that the signal is received during the predetermined time period, counting a number of times of the speech recognition failure occurring by the early utterance, re-performing speech recognition when the counted number of times is less than a reference number of times, and outputting early utterance habit guidance information when the counted number of times is the same as the reference number of times.

Abstract: Embodiments of the present disclosure provide a method and device for speech recognition. The solution comprises: receiving a first speech signal issued by a user; performing analog to digital conversion on the first speech signal to generate a first digital signal after the analog to digital conversion; extracting a first speech parameter from the first digital signal, the first speech parameter describing a speech feature of the first speech signal; if the first speech parameter coincides with a first prestored speech parameter in a sample library, executing control signalling instructed by the first digital signal, the sample library prestoring prestored speech parameters of N users, N?1. The solution can be applied in a speech recognition process and can improve the accuracy of speech recognition.

Abstract: A content alignment service may generate content synchronization information to facilitate the synchronous presentation of audio content and textual content. In some embodiments, a region of the textual content whose correspondence to the audio content is uncertain may be analyzed to determine whether the region of textual content corresponds to one or more words that are audibly presented in the audio content, or whether the region of textual content is a mismatch with respect to the audio content. In some embodiments, words in the textual content that correspond to words in the audio content are synchronously presented, while mismatched words in the textual content may be skipped to maintain synchronous presentation. Accordingly, in one example application, an audiobook is synchronized with an electronic book, so that as the electronic book is displayed, corresponding words of the audiobook are audibly presented.

Abstract: A microphone apparatus is provided. Whether an electronic signal converted from an audio signal is in compliance with a preset sound characteristic and a preset voice recognition information is determined. When the electronic signal is in compliance with the preset sound characteristic and the preset voice recognition information, an actuation control signal is outputted, so as to trigger an operation of an external circuit external to the microphone apparatus.

Abstract: Systems and methods providing for secure voice print authentication over a network are disclosed herein. During an enrollment stage, a client's voice is recorded and characteristics of the recording are used to create and store a voice print. When an enrolled client seeks access to secure information over a network, a sample voice recording is created. The sample voice recording is compared to at least one voice print. If a match is found, the client is authenticated and granted access to secure information. Systems and methods providing for a dual use voice analysis system are disclosed herein. Speech recognition is achieved by comparing characteristics of words spoken by a speaker to one or more templates of human language words. Speaker identification is achieved by comparing characteristics of a speaker's speech to one or more templates, or voice prints. The system is adapted to increase or decrease matching constraints depending on whether speaker identification or speaker recognition is desired.

Abstract: An apparatus comprising a mapper configured to map an instance of a parameter according to a first mapping to generate a first mapped instance; a remapper configured to remap the first mapped instance dependent on the frequency distribution of mapped instances to generate a remapped instance with an associated order position; and an encoder configured to encode the remapped instance dependent on an order position of the remapped instance.

Abstract: The present invention provides improvements to prior art audio codecs that generate a stereo-illusion through post-processing of a received mono signal. These improvements are accomplished by extraction of stereo-image describing parameters at the encoder side, which are transmitted and subsequently used for control of a stereo generator at the decoder side. Furthermore, the invention bridges the gap between simple pseudo-stereo methods, and current methods of true stereo-coding, by using a new form of parametric stereo coding. A stereo-balance parameter is introduced, which enables more advanced stereo modes, and in addition forms the basis of a new method of stereo-coding of spectral envelopes, of particular use in systems where guided HFR (High Frequency Reconstruction) is employed. As a special case, the application of this stereo-coding scheme in scalable HFR-based codecs is described.

Abstract: The present invention provides improvements to prior art audio codecs that generate a stereo-illusion through post-processing of a received mono signal. These improvements are accomplished by extraction of stereo-image describing parameters at the encoder side, which are transmitted and subsequently used for control of a stereo generator at the decoder side. Furthermore, the invention bridges the gap between simple pseudo-stereo methods, and current methods of true stereo-coding, by using a new form of parametric stereo coding. A stereo-balance parameter is introduced, which enables more advanced stereo modes, and in addition forms the basis of a new method of stereo-coding of spectral envelopes, of particular use in systems where guided HFR (High Frequency Reconstruction) is employed. As a special case, the application of this stereo-coding scheme in scalable HFR-based codecs is described.

Abstract: To provide a bandwidth extension method which allows reduction of computation amount in bandwidth extension and suppression of deterioration of quality in the bandwidth to be extended. In the bandwidth extension method: a low frequency bandwidth signal is transformed into a QMF domain to generate a first low frequency QMF spectrum; pitch-shifted signals are generated by applying different shifting factors on the low frequency bandwidth signal; a high frequency QMF spectrum is generated by time-stretching the pitch-shifted signals in the QMF domain; the high frequency QMF spectrum is modified; and the modified high frequency QMF spectrum is combined with the first low frequency QMF spectrum.

Abstract: A method and an apparatus for processing a temporal envelope of an audio signal, and an encoder are disclosed. When multiple temporal envelopes are solved, continuity of signal energy can be well maintained, and in addition, complexity of calculating a temporal envelope is reduced. The method includes: obtaining a high-band signal of the current frame audio signal according to the received current frame audio signal; dividing the high-band signal of the current frame signal into M subframes according to a predetermined temporal envelope quantity M, where M is an integer, M is greater than or equal to 2; calculating a temporal envelope of each of the subframes; performing windowing on the first subframe of the M subframes and the last subframe of the M subframes by using an asymmetric window function; and performing windowing on a subframe except the first subframe and the last subframe of the M subframes.

Abstract: An audio signal processing device comprises a discontinuity detector configured to determine an occurrence of a discontinuity from a sudden increase of an amplitude of decoded audio obtained by decoding the first audio packet which is received correctly after an occurrence of a packet loss, and a discontinuity corrector for correcting the discontinuity of the decoded audio.

Abstract: An apparatus for decoding an encoded audio signal including bandwidth extension control data indicating either a first harmonic bandwidth extension mode or a second non-harmonic bandwidth extension mode, includes: an input interface for receiving the encoded audio signal including the bandwidth extension control data indicating either the first harmonic bandwidth extension mode or the second non-harmonic bandwidth extension mode; a processor for decoding the audio signal using the second non-harmonic bandwidth extension mode; and a controller for controlling the processor to decode the audio signal using the second non-harmonic bandwidth extension mode, even when the bandwidth extension control data indicates the first harmonic bandwidth extension mode for the encoded signal.

Abstract: The present invention relates to audio coding systems which make use of a harmonic transposition method for high frequency reconstruction (HFR). A system and a method for generating a high frequency component of a signal from a low frequency component of the signal is described. The system comprises an analysis filter bank providing a plurality of analysis subband signals of the low frequency component of the signal. It also comprises a non-linear processing unit to generate a synthesis subband signal with a synthesis frequency by modifying the phase of a first and a second of the plurality of analysis subband signals and by combining the phase-modified analysis subband signals. Finally, it comprises a synthesis filter bank for generating the high frequency component of the signal from the synthesis subband signal.

Abstract: A gaming headset receives a plurality of audio channels comprising game audio channels and a chat audio channel during play of a particular game. The gaming headset monitors the received audio channels for predefined words that are associated with particular sounds in a data structure, and in response to detecting predefined words, filters out at least a portion of the detected predefined words from the received plurality of audio channels. The monitoring compares sounds on the received audio channels with the particular sounds in the data structure and also performs signal analysis on the audio channels during game play to detect the occurrence of the predefined words. The filtering mutes one or more of the plurality of audio channels so that the detected occurrence of the one of the predefined words is not output via speakers of the gaming headset.

Abstract: In some embodiments, a method of creating an automatic language characteristic recognition system. The method can include receiving a plurality of audio recordings. The method also can include segmenting each of the plurality of audio recordings to create a plurality of audio segments for each audio recording. The method additionally can include clustering each audio segment of the plurality of audio segments according to audio characteristics of each audio segment to form a plurality of audio segment clusters. Other embodiments are provided.

Abstract: The dynamic range and power efficiency of a voice-activated system may be improved by dynamically adjusting the configuration of the voice-activated system's input path. In one embodiment, a first portion of audio may be received through an input path of the voice-activated system having a first configuration. A characteristic of the first portion of audio may be determined and the input path may be adjusted to a second configuration based on the determined characteristic. A second portion of audio may then be received through the input path having the second configuration, and speech analysis may be performed on the second portion of audio.

Abstract: An apparatus and method for verifying an utterance based on multi-event detection information in a natural language speech recognition system. The apparatus includes a noise processor configured to process noise of an input speech signal, a feature extractor configured to extract features of speech data obtained through the noise processing, an event detector configured to detect events of the plurality of speech features occurring in the speech data using the noise-processed data and data of the extracted features, a decoder configured to perform speech recognition using a plurality of preset speech recognition models for the extracted feature data, and an utterance verifier configured to calculate confidence measurement values in units of words and sentences using information on the plurality of events detected by the event detector and a preset utterance verification model and perform utterance verification according to the calculated confidence measurement values.

Abstract: A magnetic apparatus has a media-facing surface (MFS), a pole, a top shield, a back gap and coil(s). The pole includes a yoke extension, a yoke between the yoke extension and the MFS, and a pole tip between the yoke and the MFS. The write gap is between the top shield and the pole tip. The back gap is recessed from the ABS and magnetically and physically connects the top shield to the yoke. The coil(s) energize the pole and have multiple turns. Part of a first turn is between the yoke and the top shield. Part of a second turn is recessed from the MFS and aligned with part of the yoke extension. Part of the first turn is between the part of the second turn and the MFS. The back gap is between part of the first turn and part of the second turn.

Abstract: A near-field transducer includes an enlarged portion formed of a soft plasmonic metal. A diffusion barrier is formed on one side of the enlarged portion, the diffusion barrier made of a harder material than the soft plasmonic metal. A heat sink is formed on the diffusion barrier, the heat sink made of the soft plasmonic metal. A peg is embedded in the diffusion layer so that the peg is isolated from the enlarged portion and the heat sink. The peg made of the soft plasmonic material and extends out from the diffusion layer towards a recording medium.

Abstract: A device that includes a near field transducer (NFT), the NFT having a disc and a peg, and the peg having an air bearing surface thereof; and at least one adhesion layer positioned on at least the air bearing surface of the peg, the adhesion layer including one or more of platinum (Pt), iridium (Ir), ruthenium (Ru), rhodium (Rh), palladium (Pd), yttrium (Y), chromium (Cr), nickel (Ni), and scandium (Sc).

Abstract: According to one embodiment, a magnetic recording head includes a magnetic pole, a stacked body, and a first nonmagnetic layer. The stacked body includes first magnetic layer, a second magnetic layer provided between the first magnetic layer and the magnetic pole, and an intermediate layer provided between the first magnetic layer and the second magnetic layer and being nonmagnetic. The first nonmagnetic layer is provided between the second magnetic layer and the magnetic pole. A product of a thickness and a saturation magnetic flux density of the second magnetic layer is larger than a product of a thickness and a saturation magnetic flux density of the first magnetic layer. The length of the first magnetic layer is shorter than a length of the second magnetic layer. A current flows from the second magnetic layer toward the first magnetic layer.

Abstract: In one general embodiment, a method includes forming a slot on a tape bearing surface of at least a chip having a thin film layer with a plurality of transducers therein, the slot defining a skiving edge. A second operation is performed on the tape bearing surface of at least the chip for removing a portion of the chip positioned on an opposite side of the slot as the transducers. In another general embodiment, an apparatus includes a substrate, a thin film layer on the substrate having transducers therein, and a portion of a slot extending along the substrate, the portion of the slot defining a skiving edge. A length of a tape bearing surface between the substrate and the skiving edge is in a range of about 7 to about 30 microns.

Abstract: A magnetic read head is provided, comprising a bottom magnetic shield, a first free magnetic layer, a second free magnetic layer, and a top magnetic shield, arranged from bottom to top in this order in a stacking direction from a leading side to a trailing side of the read head. A soft bias layer is positioned below the top magnetic shield and on a back side of the first free magnetic layer and second free magnetic layer. The soft bias layer is directly coupled to the top magnetic shield, and the top magnetic shield has a unidirectional anisotropy, thereby reducing the incidence of soft bias magnetization reversal and avoiding hysteresis in the transfer curve.

Abstract: A MR sensor is disclosed with an antiferromagnetic (AFM) layer recessed behind a bottom shield to reduce reader shield spacing and improve pin related noise. Above the AFM layer is an AP2/AFM coupling layer/AP1 stack that extends from an air bearing surface to the MR sensor backside. The AP2 layer is pinned by the AFM layer, and the AP1 layer serves as a reference layer to an overlying free layer during a read operation. The AP1 and AP2 layers have improved resistance to magnetization flipping because back portions thereof have a full cross-track width “w” between MR sensor sides thereby enabling greater pinning strength from the AFM layer. Front portions of the AP1/AP2 layers lie under the free layer and have a track width less than “w”. The bottom shield may have an anti-ferromagnetic coupling structure. A process flow is provided for fabricating the MR sensor.

Abstract: A heat generating component of a slider is energized at a predetermined frequency. The heat generating component changes a spacing between a medium and the slider. A temperature response proximate a media-facing surface of the slider is measured while the heating element is energized. Based on the measured temperature response, a determination is made as to whether the media-facing surface is contaminated. In response to determining that the media-facing surface is contaminated, remedial action is taken.

Abstract: A recording head includes a layer of plasmonic metal deposited on a surface of the recording head. One or more non-self-supporting layers of crystalline material are attached to the plasmonic metal, the one or more layers of crystalline materials configured to form an active region of a laser. A waveguide is configured to receive plasmons from the laser and direct the plasmons to a recording medium.

Abstract: According to one embodiment, a magnetic disk drive in the present embodiment includes a disk including tracks, each including servo sectors, a head, and a controller configured to acquire first correction data for repeatable runout occurring in a first direction, and second correction data different from the first correction data, to write the first correction data within a first permitted range in the first direction, to write the second correction data within a write permitted range including the first permitted range and a second permitted range in a second direction opposite to the first direction, to read at least one of the first correction data and the second correction data, and to correctly place the head.

Abstract: An apparatus comprises a slider and an optical waveguide formed in the slider and configured to receive light from a laser source. A near-field transducer (NFT) is formed on the slider at or near an air bearing surface (ABS) of the slider and optically coupled to the waveguide. A bolometric sensor is positioned proximate the NFT and exposed to at least some of the light. The bolometric sensor is configured to detect changes in output optical power of the laser source and contact between the slider and a magnetic recording medium.

Abstract: A three dimensional magnetic recording medium can consist of a first recording layer vertically stacked with a second recording layer. The first stacked recording layer may be tuned with at least one discrete track physically separating multiple data tracks in the first recording layer or tuned by being configured as a bit patterned media.

Abstract: A magnetic stack includes a substrate and a magnetic recording layer disposed over the substrate. The magnetic recording layer comprises magnetic crystalline grains and a segregant disposed between grain boundaries of the crystalline grains. One or both of the magnetic crystalline grains and the segregant are doped with a rare earth or transition metal dopant in an amount that provides the magnetic recording layer with a magnetic damping value, ?, between about 0.1 to about 1.

Abstract: An enlarged substrate configuration for magnetic recording media. In some embodiments, an apparatus comprises a disc shaped member of metal or glass configured as a substrate for a magnetic data recording disc for a 2½ inch form factor data storage device or a 3½ inch form factor data storage device. The disc shaped member has a top planar surface, an opposing bottom planar surface, an outermost annular sidewall that extends between the top planar surface and the bottom planar surface at an outermost diameter of nominally 67 or 97 millimeters, respectively, and an inner annular sidewall that extends between the top planar surface and the bottom planar surface to define a central aperture.

Abstract: A tape appliance includes first and second tape head modules, and a guide surface with first and second slots. The guide surface is configured such that the as the tape moves in first and reverse directions of longitudinal tape travel, the tape wraps about the guide surface, and at least two non-overlapping wrap positions of the tape engage the first slot, and at least one wrap position of the tape engages the second slot. When the tape travel is in the first longitudinal direction, the first and second tape head modules are positioned within the first and second slots, respectively, such that the first tape head module is the leading module. When the tape travel is in the reverse longitudinal direction, the first and second tape head modules are positioned within the first and second slots, respectively, such that the first tape head module remains the leading module.

Abstract: Utilities that increase the volume of media elements that can be simultaneously loaded and/or unloaded into or from a storage library, facilitate mounting of media element magazines into a storage library, and limit access to an interior of a storage library by users during operation of robotics assemblies of the storage library. One disclosed utility includes a cartridge access port for use with a storage library that has a storage container that is pivotable (e.g., swingable) between at least first and second positions. In the first position, the storage container is adjacent an opening in the housing of the library for loading and unloading of media elements by a user into or from the container via the opening. After the storage container has swung into the second position, the storage container is spaced from the opening and faces the interior of the library for access by a robotics assembly.

Abstract: A multiple disk loader apparatus includes a plurality of rods. Each rod has a pair of pins extending radially from a side of the rod. The pair of pins are spaced circumferentially around the rod with respect to each other. Each pin has a top surface wherein the top surface of a first pin is longitudinally separated from the top surface of a second pin by a predetermined gap. A rotation device is coupled to the plurality of rods. The rotation device rotates the plurality of rods individually through a respective predetermined arc and in a respective rotational direction in order to extract a disk stack from storage and to separate a disk from the stack to load into a drive.

Abstract: A fluid-dynamic bearing system comprising a stationary bearing component (12, 16, 18) and a bearing component (14, 14a) rotatable about a rotation axis, wherein, during operation of the bearing, the stationary and rotary components are separated from each other by a bearing gap (20) filled with a bearing fluid, wherein at least one fluid-dynamic radial bearing (22, 24) and at least one fluid-dynamic thrust bearing (28) or, alternatively, at least one conical fluid-dynamic bearing are arranged along the bearing gap (20), and wherein the bearing gap (20) comprises first and second open ends sealed by a first sealing gap (34) and a second sealing gap (36). The second sealing gap (36) exclusively extends normal to the rotation axis (40).