Abstract: In one aspect of the invention, a driving circuit has a PCB, a transmitter disposed on the PCB for providing an input signal, first and second transmission lines disposed on the PCB and electrically coupled to the transmitter for transmitting the input signal, and a plurality of source drivers formed in a COF architecture disposed between the PCB and the display panel. The input signal is an encoded signal including first and second Gamma reference voltages control signals, and image data. The first and second Gamma reference voltages are transmitted by the PLC technology through the first and second transmission lines, respectively. The driving circuit is implemented with differential transmission of the Gamma voltages, the image data and the control signals.

Abstract: Some embodiments enable a first electronic device (e.g., a notebook computer) to obtain state information directly from another electronic device (e.g., a smartphone) so that the first electronic device may replicate a state of content of the other computing device. This is useful when a user of an electronic device desires to switch between one device and another device such that the user may continue an activity (e.g., playing a video game) on another device without having to restart the activity. This is also useful when a user of a first electronic device attempts to replicate the state of the activity on a second device from another user such that both users may participate in the same activity. In some embodiments, a user of a device may obtain content from a server and state information from another device to replicate the state of content on the other device.

Abstract: Embodiments of the present invention disclose a display method and a display device. 2D display is performed when an input source signal is a 2D image signal, and 3D display or 2D display is performed according to requirements and preferences of a user when the input source signal is a 3D image signal. Problems of blinking and tailing which are present in an existing display device are resolved, an edge blur phenomenon of images for fast motion pictures is illuminated, the “illusion” which is formed due to a visually temporary retention of human eyes is corrected, and picture stability is effectively improved; and a crosstalk phenomenon caused by simultaneous displaying of the left-eye image and the right-eye image in an existing active 3D technology is eliminated.

Abstract: An organic light emitting display and a method for driving the same are disclosed. The organic light emitting display includes a display panel, a data driver supplying a data signal to the display panel, a memory which is positioned inside or outside the data driver and includes at least two banks, and a sensing circuit unit which measures a threshold voltage of at least one driving transistor included in the display panel and provides compensation data. The data driver separately writes and reads previous compensation data and new compensation data provided by the sensing circuit unit in the at least two banks of the memory.

Abstract: A musical instrument having a plurality of strings is provided, the strings extending along a longitudinal dimension. The instrument includes a concave fingerboard extending along the longitudinal dimension and spaced from the plurality of strings to define the action, wherein a portion of the concave fingerboard is below an adjacent surface of a body of the musical instrument.

Abstract: The present invention relates to a axial movement tuner that pulls an instrument string in any direction within hemispherical space where the string exits the tuner. The tuner is an adjusting thumbscrew that has a through borehole for string passage and a low friction bearing recessed within the thumbscrew knob, where instrument strings are installed by simply passing them through the thumbscrew bearing, on through the hole in the thumbscrew, and out through a horn shaped opening. Ball end string movement is stopped when the ball reaches the small diameter bearing hole. Tuning is accomplished by turning the thumbscrew causing axial movement of the string end ball while the bearing limits string rotation, and this axial movement of the string within the tuner is redirected by a smooth horn shaped opening to any direction within a hemispherical space defined by the tuner string exit opening.

Abstract: A mapping estimation apparatus includes a mapping adjuster. The mapping adjuster estimates mappings which correlate a plurality of subset data items with respective parts of universal set data including union of the plurality of subset data items based on the plurality of subset data items and the universal set data. The mapping adjuster estimates a mode of selecting a plurality of codomain data items from the universal set data and modes of mappings applied to the plurality of subset data items so as to have a maximum probability that data items obtained by selecting a plurality of codomain data items of which a subset of union is the universal set data from the universal set data and applying the mappings to the plurality of subset data items as domains will be respectively the plurality of codomain data items.

Abstract: Music recognition is carried out by accepting a musical score of musical elements in a digital format, transforming the digital format into a composite musical data object that models the musical score, defining the key signatures in the composite musical data object probabilistically, computing start times to play musical elements in respective measures of the composite musical data object without regard to rhythmic values of other musical elements in the respective measures, and generating an output including the defined key signatures and computed start times.

Abstract: A sound effect data generating apparatus has a starting point setting portion, an endpoint setting portion, a travel point defining portion and a data generating portion. The starting point setting portion and the endpoint setting portion set a starting point and an endpoint specified by position information, respectively. The travel point defining portion sequentially defines a point which is situated on a line connecting between the set starting point and the set endpoint and is specified by position information, as a travel point in accordance with progression of reproduced tone signals. The data generating portion determines a value of at least one parameter on one sound effect which is to be added to the tone signals on the basis of the position information of the defined travel point, the set starting point and the set endpoint, and generates sound effect data in accordance with the determined parameter value.

Abstract: A resonance tone generation apparatus 20 is applied to an electronic musical instrument DM having a tone generator for generating a musical tone signal indicative of a piano sound in accordance with a tone generation instruction signal having a key number n, and a plurality of output portions for externally outputting a musical tone signal. The resonance tone generation apparatus 20 has a plurality of resonance tone generation circuits 30(n) each of which is assigned a different key number n, retrieves a musical tone signal indicative of a musical sound of the piano, generates a musical tone signal indicative of a resonance tone imitating a tone of a string of the piano resonated by the musical sound of the piano indicated by the retrieved musical tone signal, and supplies the generated musical tone signal to the plurality of output portions.

Abstract: A sub-harmonic mixer comprising first and second input ports, and an output port, and arranged to output products of signals applied at the input ports. The sub-harmonic mixer comprises a first stage a second stage arranged in series with each other, with the input ports of the sub-harmonic mixer being input ports of the first stage, and the output port of the sub-harmonic mixer being an output port of the second stage. An output port of the first stage is connected to an input port of the second stage. The first stage is arranged to generate a fundamental product and a first third order intermodulation distortion product in-phase with each other. The second stage is arranged to generate a second third order intermodulation distortion product in anti-phase to the fundamental product generated by the second stage.

Abstract: Example methods and systems for determining media files based on activity levels are described. An example method includes receiving information indicative of a first speed of the computing device, and receiving information indicative of a geography of a location of the computing device. The method further includes determining, from a plurality of media files tagged with respective tempo identifiers, a first media file based on the geography of the location of the computing device and also having a tempo that substantially matches to the first speed of the computing device. The method includes providing an indication of the first media file to a media player, and based on a change in the first speed of the computing device to a second speed, determining from the plurality of media files tagged with respective tempo identifiers, a second media file having a tempo that substantially matches to the second speed.

Abstract: An electric instrument music control device is provided having a foot pedal comprising a base portion and a treadle, wherein the treadle moves with respect to the base portion. The device further has a magnetic displacement sensor coupled to the base portion and a magnet coupled to the treadle. The magnet is located adjacent the magnetic displacement sensor to place the sensor in a field-saturated mode, wherein the magnet moves with respect to the magnetic displacement sensor in response to movement of the treadle with respect to the base portion. A sound characteristic of the electric instrument is modified in response to moving the magnet with respect to the magnetic displacement sensor.

Abstract: A reversing configuration control for string instruments connects to a pair of pickup coils located on a string instrument for selectively electrically configuring and blending the outputs of the pickup coils. The reversing configuration control includes a pair of potentiometers mechanically coupled for concurrent mechanical travel of the respective displaceable contacts thereof. The pair of potentiometers are operatively coupled to the pair of pickup coils and a pair of output terminals to vary the electrical configuration of the pair of pickup coils between (a) the pair of pickup coils being coupled with like polarity in one of series or parallel with respect to the output terminals, or (b) effectively coupling only one of the pair of pickup coils to the output terminals, or (c) the pair of pickup coils being coupled with opposing polarity in one of series or parallel with respect to the output terminals.

Abstract: An acoustic device includes an opening; a flange forming the opening; a first member including the opening and the flange; and a second member joined to the first member, thereby forming a cavity. The second member is formed of a material with a density lower than a material of the first member.

Abstract: An active noise control system and method for tuning an acoustic noise signal at a listening position are disclosed in which a first weighting element is connected in the filter coefficient copy path and/or a second weighting element is connected in the microphone path.

Abstract: In accordance with systems and methods of this disclosure, a method may include generating a feedforward anti-noise signal component from a result of measuring with the reference microphone countering the effects of ambient audio sounds at an acoustic output of a transducer by filtering an output of the reference microphone, adaptively generating a feedback anti-noise signal component from a result of measuring with an error microphone for countering the effects of ambient audio sounds at the acoustic output of the transducer by adapting a response of a feedback adaptive filter that filters a synthesized reference feedback to minimize the ambient audio sounds in the error microphone signal, wherein the synthesized reference feedback is based on a difference between the error microphone signal and the feedback anti-noise signal component.

Abstract: An image forming apparatus includes a driving force transmitting mechanism transmitting a driving force from a driving source to a rotating member through a plurality of coupled driving force transmitting members. The driving force transmitting mechanism has a supporting member and a drive case. The support member supports the driving source and the rotating members. The drive case is provided on the support member and forms a closed space with the supporting member. The closed space stores the driving force transmitting members. The drive case is formed with a sound insulating part. The sound insulating part has at least either one of concave and convex parts at a position closest to the coupling part of the driving force transmitting members.

Abstract: In accordance with embodiments of the present disclosure, an adjustable equalization filter may have a response that generates an equalized source audio signal from a source audio signal to account for effects of changes in an electro-acoustical path of the source audio signal to a transducer. An equalizer coefficient control block may adapt the response of the adjustable equalization filter in response to changes in a response of a secondary path estimate filter for modeling the electro-acoustical path of a source audio signal through the transducer, wherein a response of the secondary path estimate filter is adapted in conformity with an error microphone signal indicative of the acoustic output of the transducer.

Abstract: An acoustic wave device built-in module includes: a multilayer wiring board formed by stacking an insulating layer and a wiring layer; an acoustic wave device embedded in the multilayer wiring board; and an electronic component located on the multilayer wiring board and electrically coupled to the acoustic wave device, wherein the acoustic wave device includes: an electrode that is located on a substrate and excites an acoustic wave; and a sealing portion that includes a frame body located on the substrate so as to surround the electrode and a lid located on the frame body so as to form an air space above the electrode, and the lid is recessed toward the substrate.

Abstract: The invention relates to a device and method for generating noise for a motor vehicle, having at least one sensing unit (101), which is designed to measure at least one operating signal (11) of the motor vehicle and to forward a basic signal (12) on the basis thereof to a tuning unit (102), wherein the tuning unit (102) is designed to convert the basic signal (12) by means of filtering and acoustic shaping to form a noise signal (13) and to forward the noise signal (13) to an amplifier unit (103), wherein the amplifier unit (103) is designed to amplify the noise signal (13) and to forward an amplified noise signal (14) to a converter unit (104), and wherein the converter unit (104) is designed to convert the amplified noise signal (14) into corresponding acoustic signals in order to generate noise.

Abstract: Architecture for playing back personal text-based messages such as email and voicemail over a vehicle-based media system. The user can use a cell phone that registers over a cellular network to an IMS (IP multimedia subsystem) to obtain an associated IP address. The personal messages are then converted into audio signals using a remote text-to-voice (TTV) converter and transmitted to the phone based on the IP address. The phone then transmits the audio signals to the vehicle media system for playback using an unlicensed wireless technology (e.g., Bluetooth, Wi-Fi, etc.). Other alternative embodiments include transmitting converted message directly to the media system, via a satellite channel, converting the messages via a TTV converter on the cell phone, and streaming the converted messages to the phone and/or the media system for playback.

Abstract: A method for speech recognition is implemented in the specific form of computer processes that function in a computer processor. That is, one or more computer processes: process a speech input to produce a sequence of representative speech vectors and perform multiple recognition passes to determine a recognition output corresponding to the speech input. At least one generic recognition pass is based on a generic speech recognition arrangement using generic modeling of a broad general class of input speech. And at least one adapted recognition pass is based on a speech adapted arrangement using pre-adapted modeling of a specific sub-class of the general class of input speech.

Abstract: A method and system for improving the accuracy of a speech recognition system using word confidence score (WCS) processing is introduced. Parameters in a decoder are selected to minimize a weighted total error rate, such that deletion errors are weighted more heavily than substitution and insertion errors. The occurrence distribution in WCS is different depending on whether the word was correctly identified and based on the type of error. This is used to determine thresholds in WCS for insertion and substitution errors. By processing the hypothetical word (HYP) (output of the decoder), a mHYP (modified HYP) is determined. In some circumstances, depending on the WCS's value in relation to insertion and substitution threshold values, mHYP is set equal to: null, a substituted HYP, or HYP.

Abstract: Disclosed are techniques and systems to provide a narration of a text. In some aspects, the techniques and systems described herein include generating a timing file that includes elapsed time information for expected portions of text that provides an elapsed time period from a reference time in an audio recording to each portion of text in recognized portions of text.

Abstract: Provided are a frame error concealment method and apparatus and an error concealment scheme construction method and apparatus. The frame error concealment method includes generating a new signal by synthesizing a plurality of previous signals that are similar to a signal of an error frame and reconstructing the signal of the error frame using the generated signal.

Abstract: A method is provided for concealing a lost audio frame of a received audio signal by performing a sinusoidal analysis of a part of a previously received or reconstructed audio signal. The sinusoidal analysis involves identifying frequencies of sinusoidal components of the audio signal, and applying a sinusoidal model on a segment of the previously received or reconstructed audio signal. The segment is used as a prototype frame in order to create a substitution frame for a lost audio frame. The method includes creating the substitution frame for the lost audio frame by time-evolving sinusoidal components of the prototype frame, up to the time instance of the lost audio frame, in response to the corresponding identified frequencies. The method further includes performing at least one of an enhanced frequency estimation and an adaptation of the creating of the substitution frame in response to the tonality of the audio signal.

Abstract: A frame error concealment method based on frames including transform coefficient vectors including the following steps: It tracks (S11) sign changes between corresponding transform coefficients of predetermined sub-vectors of consecutive good stationary frames. It accumulates (S12) the number of sign changes in corresponding sub-vectors of a predetermined number of consecutive good stationary frames. It reconstructs (S13) an erroneous frame with the latest good stationary frame, but with reversed signs of transform coefficients in sub-vectors having an accumulated number of sign changes that exceeds a predetermined threshold.

Abstract: Provided are a multi-channel audio down-mixing method and apparatus for selecting down-mix target channels based on a calculation of correlations between channels and then down-mixing the down-mix target channels. The method includes: calculating correlations between channels of multi-channel audio; selecting a first channel and a second channel, among the channels of the multi-channel audio, that are to be down-mixed, based on the calculated correlations; and down-mixing the selected first channel and the selected second channel.

Abstract: An audio processing system (100) comprises a front-end component (102, 103), which receives quantized spectral components and performs an inverse quantization, yielding a time-domain representation of an intermediate signal. The audio processing system further comprises a frequency-domain processing stage (104, 105, 106, 107, 108), configured to provide a time-domain representation of a processed audio signal, and a sample rate converter (109), providing a reconstructed audio signal sampled at a target sampling frequency. The respective internal sampling rates of the time-domain representation of the intermediate audio signal and of the time-domain representation of the processed audio signal are equal. In particular embodiments, the processing stage comprises a parametric upmix stage which is operable in at least two different modes and is associated with a delay stage that ensures constant total delay.

Abstract: Systems, methods, and apparatus for a unified approach to encoding different types of audio inputs are described.

Abstract: Provided is a method and apparatus for encoding/decoding an audio signal. Sections which are not used to output noise components near important spectral components and sub-bands which are not used to output noise components, are determined to be encoded or decoded, so that the efficiency of encoding and decoding an audio signal increases, and sound quality can be improved using less bits.

Abstract: Provided are a method and apparatus for encoding and decoding a high frequency signal by using a low frequency signal. The high frequency signal can be encoded by extracting a coefficient by linear predicting a high frequency signal, and encoding the coefficient, generating a signal by using the extracted coefficient and a low frequency signal, and encoding the high frequency signal by calculating a ratio between the high frequency signal and an energy value of the generated signal. Also, the high frequency signal can be decoded by decoding a coefficient, which is extracted by linear predicting a high frequency signal, and a low frequency signal, and generating a signal by using the decoded coefficient and the decoded low frequency signal, and adjusting the generated signal by decoding a ratio between the generated signal and an energy value of the high frequency signal.

Abstract: An encoder (1201) for encoding a plurality of audio signals comprises a selector (1303) which selects a subset of time-frequency tiles to be downmixed and a subset of tiles to be non-downmix. A downmix indication is generated which indicates whether tiles are encoded as downmixed encoded tiles or as non-downmix tiles. An encoded signal comprising the encoded tiles and the downmix indication is fed to a decoder (1203) which includes a receiver (1401) for receiving the signal. A generator (1403) generates output signals from the encoded time-frequency tiles where the generation of the output signals includes an upmixing for tiles that are indicated by the downmix indication to be encoded downmixed tiles. The invention may provide more flexible and/or improved encoding/decoding and may specifically provide improved scalability, especially at higher data rates.

Abstract: Loud sounds with fast rise times, like gunfire and explosions, can cause noise-induced hearing loss (NIHL). Unfortunately, current models do not adequately explain how impulsive sounds cause NIHL, which makes it difficult to predict and prevent NIHL on battlefields and other hostile or rugged environments. Fortunately, the impulsive sounds experienced by soldiers and others working in rugged environments can be recorded using a compact, portable system that acquires, digitizes, and stores high-bandwidth audio data. An example of this system can be mounted on a helmet or other article and used to record hours of audio data at a bandwidth of 20 kHz or higher, which is broad enough to capture sounds with rise times less than 50 ms. An analog-to-digital converter (ADC) digitizes these broadband audio signals at rate of 40 kHz or higher to preserve the impulse information. A processor transfers the digitized samples from a buffer to a memory card for later retrieval using an interrupt-driven processing technique.

Abstract: A speech processing apparatus that collects sound signals. With each of the collected sound signals, the apparatus may estimate a direction of a sound source, and select an extension filter that is applied to each sound signal. The extension filter may correspond to the estimated sound source of each of the sound signals. In addition, each of the sound signals may be corrected using the extension filter, and a reverberation reduction of the corrected sound signals and the collected sound signals may be performed.

Abstract: Methods and systems provide a partitioned IP core and hierarchical power management to reduce power consumption and footprint size of an “always-on” pulse density modulation (PDM) sensor system. The IP core may be partitioned into a register transfer level (RTL) block and a firmware block. The RTL may include a first stage decimation filter, storage, and, optionally, a sound energy detector. The firmware block may include subsequent decimation filter(s) and sensor processing logic, e.g., a sound trigger algorithm. In operation, the firmware block may conserve energy by being in a power-off or power-saving mode by default. Responsive to a trigger by the sound energy detector, the firmware block may wake up, receive data from the RTL block, and process the data. The sound energy detector may output the trigger based on characteristics of the received sample such as signal strength, noise strength, and type.

Abstract: According to an embodiment, a signal processing apparatus includes an ambient sound estimating unit, a representative component estimating unit, a voice estimating unit, and a filter generating unit. The ambient sound estimating unit is configured to estimate, from the feature, an ambient sound component that is non-stationary among ambient sound components having a feature. The representative component estimating unit is configured to estimate a representative component representing ambient sound components estimated from one or more features for a time period, based on a largest value among the ambient sound components within the time period. The voice estimating unit is configured to estimate, from the feature, a voice component having the feature. The filter generating unit is configured to generate a filter for extracting a voice component and an ambient sound component from the feature, based on the voice component and the representative component.

Abstract: A conferencing system and method involves conducting a conference between endpoints. The conference can be a videoconference in which audio data and video data are exchanged or can be an audio-only conference. Audio of the conference is obtained from one of the endpoints, and speech is detected in the obtained audio. The detected speech is analyzed to determine that the detected speech constitutes a speech fragment, and an indicia indicative of the determined speech fragment is generated. For a videoconference, the indicia can be a visual cue to be added to video for the given endpoint when displayed at other endpoints. For an audio-only conference, the indicia can be an audio cue to be added to the audio of the conference at the other end points.

Abstract: At a microphone, voice activity is detected in a data stream while simultaneously buffering audio data from the data stream to create buffered data. A signal is sent to a host indicating the positive detection of voice activity in the data stream. When an external clock signal is received from the host, the internal operation of the microphone is synchronized with the external clock signal. Buffered data stream is selectively sent through a first path, the first path including a buffer having a buffer delay time representing the time the first data stream takes to move through the buffer. The data stream is continuously sent through a second path as a real-time data stream, the second path not including the buffer, the real-time data stream beginning with the extended buffer data at a given instant in time. The buffered data stream and the real-time data stream are multiplexed onto a single data line and transmitting the multiplexed data stream to the host.

Abstract: Up-sampler generates an up-sampled sound signal from the sound signal. From the up-sampled sound signal, odd-ordered high-harmonic generator generates an odd-ordered high-harmonic, and even-ordered high-harmonic generator generates an even-ordered high-harmonic. Vowel sound detector identifies whether or not the sound signal is vowel sound, and generates a first gain value and a second gain value. First gain controller amplifies or attenuates the odd-ordered high-harmonic based on the first gain value, and outputs the resultant odd-ordered high-harmonic. Second gain controller amplifies or attenuates the even-ordered high-harmonic based on the second gain value, and outputs the resultant even-ordered high-harmonic. Sound signal processing device adds the gain-adjusted odd-ordered high-harmonic and the gain-adjusted even-ordered high-harmonic to the up-sampled sound signal, and outputs the up-sampled sound signal having the high-harmonics added.

Abstract: A main pole of a perpendicular magnetic recording write head is disclosed with a non-rectangular shape at the air bearing surface and a rectangular shape behind the air bearing surface. The main pole transitions from the non-rectangular shape to a rectangular shape at a distance of 10 nm to 100 nm behind the air bearing surface. The main pole includes leading and trailing beveled surfaces that extend from the non-rectangular shape of the pole at the air bearing surface toward the rectangular shape transition.

Abstract: A data storage device may be tested during or after manufacture by a testing device that may have at least a work piece with at least one contact pad concurrently contacting bottom and sidewall surfaces of a probe tip with a centering feature of the at least one contact pad.

Abstract: A magneto-resistive effect element (MR element) has a first shield layer; a second shield layer; an inner shield layer that is positioned between the first shield layer and the second shield layer, and that makes contact with the first shield layer and faces the air bearing surface (ABS); and a multilayer film that is positioned between the first shield layer and the second shield layer. The multilayer film has a free layer; a first pinned layer; a nonmagnetic spacer layer; a second pinned layer that fixes the magnetization direction of the first pinned layer; and an antiferromagnetic layer that is exchange-coupled with the second pinned layer. The antiferromagnetic layer faces the back surface of the inner shield layer viewed from the ABS. The MR element has an insulating layer positioned between the antiferromagnetic layer and the inner shield layer.

Abstract: Implementations disclosed herein include a reader comprising a magnetically free layer and first barrier layer, wherein the barrier layer is in direct contact with a bottom shield in a down-track direction. Another implementation includes a device comprising a sensor stack comprising a free layer and a barrier layer; a synthetic antiferromagnetic shield layer comprising a reference layer and a pinned layer, wherein direction of magnetization of the reference layer forms an obtuse angle with direction of magnetization of the free layer in a quiescent state.

Abstract: A lateral spin valve reader that includes a detector structure located proximate to a bearing surface and a spin injection structure located away from the bearing surface. The lateral spin valve reader also includes a channel layer extending from the detector structure to the spin injection structure. An exterior cladding, disposed around the channel layer, suppresses spin-scattering at surfaces of the channel layer.

Abstract: A thermally-assisted magnetic recording head includes a main pole, a waveguide, and a plasmon generator. The waveguide includes a core and a cladding. The main pole and the core are located on the same side in the direction of travel of a recording medium relative to the plasmon generator. The main pole has a first end face located in the medium facing surface, and a second end face opposite to the first end face. The core has a front end face opposed to the second end face of the main pole. The cladding includes an interposition section interposed between the front end face of the core and the second end face of the main pole. The front end face of the core and the second end face of the main pole are inclined with respect to the medium facing surface.

Abstract: A magnetic write apparatus has a media-facing surface (MFS), a pole, a trailing shield, coil(s) and a write gap between the pole and trailing shield. The pole includes leading and trailing surfaces, a main portion having a first bevel and an additional portion having a second bevel and adjoining the main portion. The first bevel adjoins the MFS and is at a nonzero, acute first bevel angle from a direction perpendicular to the MFS. The second bevel is recessed from the MFS by not more than eighty nanometers, oriented at a second bevel angle from the direction and offset from the first bevel by a taper having a taper angle that is greater than the first bevel angle. The additional portion is at least ten nanometers and not more than sixty nanometers thick as measured from the first bevel.

Abstract: The magnetic recording medium for a heat-assisted recording system has a magnetic recording layer on a non-magnetic substrate and a protective layer on top of the magnetic recording layer. The protective layer includes a first lower protective layer on top of the magnetic recording layer, a first upper protective layer on the first lower protective layer, and a second protective layer on the first upper protective layer. The first lower protective layer is composed mainly of an element selected from the group consisting of Si, Al and Cu, and the first upper protective layer is a layer configured by an oxide of the material of the first lower protective layer.

Abstract: A method of protecting a magnetic information storage medium is described. The method includes fabricating a film over a surface of the magnetic information storage medium. The film includes an amorphous, uniform, homogeneous solid solution of carbon, hydrogen, silicon, and oxygen. A magnetic storage medium with such a protective film is described.