Abstract: Methods and systems involving the orienting of video data based on the orientation of a display are described herein. An example system may be configured to (1) receive first video data, the first video data corresponding to a first orientation of the image-capture device; (2) send the first video data to a second computing device; (3) receive, from the second computing device, first orientation data indicating a requested orientation of the image-capture device; (4) cause a visual depiction of the requested orientation to be displayed on a graphical display; (5) receive second video data, the second video data corresponding to a second orientation of the image-capture device, where the second orientation is closer to the requested orientation than is the first orientation; and (6) send the second video data to the second computing device.

Abstract: A vehicular display device includes: a display means; a control means for controlling the display of the display means; a non-volatile memory for storing at least an operation program for the control means or image data to be displayed on the display means; and a third connection terminal that can be connected with either a first connection terminal or a second connection terminal having different terminal arrangements. The vehicular display device, housed in an instrument panel of a vehicle, is provided with an external device that communicates with the control means by connecting the first connection terminal or the second connection terminal to the third connection terminal, wherein the control means determines whether over-writing of the non-volatile memory should be allowed in accordance with the type of the connection terminal connected to the third connection terminal.

Abstract: A method for updating images displayed on an electronic display of a mobile device includes determining a time interval between successive updates of an image frame. The image frame is displayed on the electronic display, and the mobile device is configured to be in a low power mode between the successive updates. The method also includes receiving one or more application updates within the time interval after a first update of the image frame, and updating the image frame after the time interval based on the one or more application updates.

Abstract: A method and device for compressing and decompressing data information, a drive compensation method and device, and a display device. The method for compressing data information includes: acquiring data information corresponding to a sub pixel unit; establishing a distribution function model according to the data information; obtaining a valid option value section according to the distribution function model and a valid threshold value; and dividing the valid option value section into N compression sections, and compressing data information corresponding to each of the compression sections to M times of data information corresponding to all the sub pixel units according to a storage length P of the data information corresponding to the sub pixel unit to obtain N compressed data information blocks.

Abstract: Techniques are disclosed for video playback decoding surface prediction. For instance, in some embodiments, video content may be parsed for information that can be used to predict what surfaces (e.g., computer graphics shapes to be rendered, as defined by vertices specifying the location and possibly other attributes of the shape) are most likely to be accessed, for example, by a display or a graphics processing unit (GPU) in the near future. In accordance with some embodiments, these surfaces may be pre-loaded, for example, into cache memory or other desired high-bandwidth memory in advance to minimize or otherwise reduce memory access latency. In some cases, these surfaces may be entered in a list that is kept updated with each new input frame, and the surfaces in that list may be kept inside the cache (or other high-bandwidth memory) for future display or GPU access.

Abstract: A universal tuning knob for stringed instruments is presented. The invention consists of a knob with an elongated and threaded cylindrical shaft. The shaft includes a lumen sized to receive the machine head post of any stringed instrument. The outside perimeter of the knob is configured for grip. The threaded cylindrical shaft includes a slot on its side wall to receive and retain a bushing. The bushing includes a second slot that can be notched (i.e. sized) to fit and lock the machine head post of any stringed instrument. A sleeve lock with internal threads is threaded onto the threaded cylindrical shaft of the knob thereby retaining the bushing. An optional second lumen on the head of the knob is configured to receive a mounting screw.

Abstract: A plectrum for playing a stringed musical instrument includes a body with two opposing sides, a lower plucking region, a central gripping region and a top region having a transverse groove. A wedge shaped piece of material is secured into the transverse groove on the top of the plectrum to provide a second surface for impinging or actuating the strings of the instrument. Typically, the body of the plectrum is composed of a semi rigid or flexible plastic, while the wedge of material is composed of a harder substance, such as glass or polished stone.

Abstract: Methods for using at least one musical instrument mute for electronic modification of sound emitted from a musical instrument. A mute body is positionable within a bell or horn of the musical instrument, and the body has a proximal end portion and is configured to at least partially occlude the bell or horn. A microphone is positioned at the proximal end portion of the mute body and is configured to transduce a sound produced by the musical instrument. A speaker is positioned in the mute body as well. A microcontroller is configured to receive a signal from the microphone and to electronically modify the sound of the instrument when emitted through the speaker. Some mutes also provide a communication transceiver, sensors, and input devices to remotely control and manipulate sound produced by the mute.

Abstract: Drum pedal assemblies are disclosed which can include one or more adjustment features. Adjustment features which can be included in embodiments of the present disclosure can include components for adjusting pedal angle and/or height, and the resistance offered by the pedal (such as due to a spring connection). Pedal assemblies according to some embodiments of the present disclosure can include rotatable base plates for easy storage and travel. Pedal assemblies according to some embodiments the present disclosure can include securing means such as retractable spikes for stabilizing the pedal assembly.

Abstract: A method is described comprising receiving a stem signal and an audio mix signal, wherein the audio mix signal comprises information of the stem signal. The method includes applying a first transform to the stem signal to provide a first stem spectrum, applying a second transform to the stem signal to provide a second stem spectrum, generating a plurality of mix signals using the audio mix signal, applying a first transform to each mix signal of the plurality of mix signals to provide a corresponding first mix signal spectrum, applying a second transform to each mix signal of the plurality of mix signals to provide a corresponding second mix signal spectrum, and using information of the first stem spectrum, the second stem spectrum, a first mix signal spectrum, or a second mix signal spectrum to detect the information of the stem signal in the audio mix signal.

Abstract: Provided is an electronic percussion instrument that is capable of simulating a playing technique for an acoustic percussion instrument. A tubular body part is opened on an axial end surface, and a head is attached to the axial end surface to be struck on the front surface. A capacitance sensor includes an electrode that generates a capacitance with respect to a detected conductor, such as a human body, positioned on the front surface side of the head. Because the capacitance sensor detects a change of a capacitance corresponding to a distance between the electrode and the detected conductor, whether the detected conductor approaches (contacts) the head or presses the head can be determined. As a result, the playing technique for the acoustic percussion instrument is simulated.

Abstract: A system and method use an array of ultrasonic transducers to emit and receive sound in a phased array fashion by using acoustic waveguides to achieve a desired acoustic radiation and reception pattern. A chip package attached to an acoustic transducer array includes acoustic waveguides coupled to acoustic ports. Each waveguide is coupled between a corresponding acoustic transducer and a corresponding acoustic port. A spacing of a pair of acoustic ports is different than a spacing of a corresponding pair of acoustic transducers.

Abstract: An audio customization system responsive to one or more inputs that enhance aspects of an audio output and one or more inputs that diminish aspects of an audio output. The system is set up to be able to lessen the influence of ambient audio or in some situations enhance ambient audio over source audio. The system may specify aspects of audio to be modified by specification of filtering algorithm, characterization of audio samples, monitored distortion, user selection, location specification or environmental specification.

Abstract: A flexible waveguide band is provided for that may be worn by a user around their neck, for example. The flexible waveguide band includes a spine, such as a pliable rod; a plurality of vertebrae mounted on the spine, with each vertebra including at least one waveguide channel guide; and an encapsulating potting material. The flexible waveguide band may also be attached to an acoustic pod, including an acoustic driver that may radiate sound waves into said waveguide channels and exit via a sound outlet opening in the flexible band.

Abstract: Systems, methods, and computer-readable storage media for providing for intelligent switching of languages and/or pronunciations in a text-to-speech system. As the system receives text, the text is analyzed to identify portions which should have speech constructed using a pronunciation distinct from the remaining portions of the text. The text-to-speech system uses multiple pronunciation dictionaries to generate and produce speech corresponding to the text, where the identified portions of the text are in a different language or have a different accent from the remainder of the text. Having generated speech corresponding to the text in multiple languages, accents, or dialects, the system combines the portions, then communicates the speech to the text recipient.

Abstract: A voice processing device includes a voice recognition unit for extracting a command contained in an input voice, and a control unit for controlling operation of the voice processing device itself and/or instructing external devices to operate, based on the extracted command and/or operation instruction information provided. The control unit stores, in a storage device as an operation history, operation information associating the command or a part thereof and/or the operation instruction information or a part of the operation instruction information with details of the operation which is performed by the voice processing device and/or the external device based on the command and/or the operation instruction information. When a voice is input anew, the control unit extracts from the operation history an operation record containing a command or a part of a command, and controls operation of the voice processing device itself and/or instructs an external device to operate.

Abstract: A system and method of processing disfluent speech at an automatic speech recognition (ASR) system includes: receiving speech from a speaker via a microphone; determining the received speech includes disfluent speech; accessing a disfluent speech grammar or acoustic model in response to the determination; and processing the received speech using the disfluent speech grammar.

Abstract: A system that is capable of controlling multiple entertainment systems and/or speakers using voice commands. The system receives voice commands and may determine audio sources and speakers indicated by the voice commands. The system may generate audio data from the audio sources and may send the audio data to the speakers using multiple interfaces. For example, the system may send the audio data directly to the speakers using a network address, may send the audio data to the speakers via a voice-enabled device or may send the audio data to the speakers via a speaker controller. The system may generate output zones including multiple speakers and may associate input devices with speakers within the output zones. For example, the system may receive a voice command from an input device in an output zone and may reduce output audio generated by speakers in the output zone.

Abstract: A method performing voice based actions by an electronic device includes receiving a voice command from a user; determining a relationship between the voice command and a context a historic voice command of the user, and performing an action by executing the voice command based on the context of the historic voice command.

Abstract: A computer system may receive, from an acoustic transducer and/or a portable electronic device, information specifying one or more desired operations that are to be performed. This information may include natural language describing the one or more desired operations. In response, the computer system may transform the natural language into the one or more desired operations, and may identify a subset of the electronic devices based on the one or more desired operations and a predetermined supervised learning model that relates desired operations and the electronic devices. Next, the computer system may determine a set of commands based on the one or more desired operations and the subset of the electronic devices, and the computer system may provide the set of commands to the identified subset of the electronic devices, which may perform the one or more desired operations.

Abstract: Methods and systems are described herein related to enabling service providers to address voice-activated commands. An example method may involve: receiving a first utterance on a computing device, where the first utterance includes a first command; selecting a service action corresponding to the first command; determining a selected service provider for the selected service action, where the selected service provider is selected from a plurality of service providers; and sending a service fulfillment request to the selected service provider to execute the selected service action.

Abstract: Disclosed herein is method of performing speech recognition using audio and visual information, where the visual information provides data related to a person's face. Image preprocessing identifies regions of interest, which is then combined with the audio data before being processed by a speech recognition engine.

Abstract: Contextual note taking is described. A note taking assistant can receive an indication of a specific presentation session. This indication can be used by the note taking assistant to access information or content related to the session. The note taking assistant can receive specific presentation session content, which includes identifiable context images. Identifiable context images are meant to define an individual page, an individual slide, or other atomic unit in the presentation. The note taking assistant operates by receiving a navigation message, changing the current assistant context image to a current presenter context image based on the navigation message; receiving a speech-to-text message comprising a unit of text; displaying the current presenter context image, and displaying the unit of text associated with the current presenter context image; and storing the unit of text associated with the current presenter context image.

Abstract: Disclosed herein is a GPU-accelerated speech recognition engine optimized for faster than real time speech recognition on a scalable server-client heterogeneous CPU-GPU architecture, which is specifically optimized to simultaneously decode multiple users in real-time. In order to efficiently support real-time speech recognition for multiple users, a “producer/consumer” design pattern is applied to decouple speech processes that run at different rates in order to handle multiple processes at the same time. Furthermore, the speech recognition process is divided into multiple consumers in order to maximize hardware utilization. As a result, the platform architecture is able to process more than 45 real-time audio streams with an average latency of less than 0.3 seconds using one-million-word vocabulary language models.

Abstract: An electronic device is provided. The electronic device includes a processor configured to perform automatic speech recognition (ASR) on a speech input by using a speech recognition model that is stored in a memory and a communication module configured to provide the speech input to a server and receive a speech instruction, which corresponds to the speech input, from the server. The electronic device may perform different operations according to a confidence score of a result of the ASR. Besides, it may be permissible to prepare other various embodiments speculated through the specification.

Abstract: In one aspect, a method includes receiving a prompt for use with text-dependent speaker verification; generating a linguistic representation of the prompt, wherein the linguistic representation comprises a sequence of speech units; obtaining a plurality of feature vectors or a plurality of acoustic models; generating a universal background model for the prompt using the plurality of feature vectors or the plurality of acoustic models; receiving audio enrollment data of a first speaker speaking the prompt; and creating a first speaker verification model for the first speaker by adapting the universal background model using the audio enrollment data.

Abstract: A device includes an encoder and a transmitter. The encoder is configured to determine a mismatch value indicative of an amount of temporal mismatch between a reference channel and a target channel. The encoder is also configured to determine whether to perform a first temporal-shift operation on the target channel at least based on the mismatch value and a coding mode to generate an adjusted target channel. The encoder is further configured to perform a first transform operation on the reference channel to generate a frequency-domain reference channel and perform a second transform operation on the adjusted target channel to generate a frequency-domain adjusted target channel. The encoder is also configured to estimate one or more stereo cues based on the frequency-domain reference channel and the frequency-domain adjusted target channel. The transmitter is configured to transmit the one or more stereo cues to a receiver.

Abstract: A device includes an encoder and a transmitter. The encoder is configured to generate a first high-band portion of a first signal based on a left signal and a right signal. The encoder is also configured to generate a set of adjustment gain parameters based on a high-band non-reference signal. The high-band non-reference signal corresponds to one of a left high-band portion of the left signal or a right high-band portion of the right signal as a high-band non-reference signal. The transmitter is configured to transmit information corresponding to the first high-band portion of the first signal. The transmitter is also configured to transmit the set of adjustment gain parameters corresponding to the high-band non-reference signal.

Abstract: A method performed by an audio decoder for reconstructing N audio channels from an audio signal containing M audio channels is disclosed. The method includes receiving a bitstream containing an encoded audio signal having M audio channels and a set of spatial parameters, the set of spatial parameters including an inter-channel intensity difference parameter and an inter-channel coherence parameter. The encoded audio bitstream is then decoded to obtain a decoded frequency domain representation of the M audio channels, and at least a portion of the frequency domain representation is decorrelated with an all-pass filter having a fractional delay. The all-pass filter is attenuated at locations of a transient. A matrixed version of the decorrelated signals are summed with a matrixed version of the decoded frequency domain representation to obtain N audio signals that collectively having N audio channels where M is less than N.

Abstract: Systems and methods to determine the status of transmitted audio packets. The audio packets can include digital components that are presented by a client computing device. Based on presentation variables and determinations of presentation status, the system can generate a predictive model that can be used to dynamically determines presentation values of audio packets prior to their transmission to a client computing device.

Abstract: A waveform correction processor corrects the waveform of a first digital audio signal (a CD signal, for example) having a first sampling frequency. A bit depth and sampling frequency converter converts the first digital audio signal with the waveform corrected by the first waveform correction processor to a second digital audio signal (a high-resolution digital audio signal, for example) having a second sampling frequency, which is higher than the first sampling frequency. The waveform correction processor corrects the waveform of the second digital audio signal.

Abstract: A method of enhancing speech quality includes: generating a high-frequency signal by using a low-frequency signal in a time domain; combining the low-frequency signal with the high-frequency signal; transforming the combined signal into a spectrum in a frequency domain; determining a class of a decoded speech signal; predicting an envelope from a low-frequency spectrum obtained in the transforming; and generating a final high-frequency spectrum by applying the predicted envelope to a high-frequency spectrum obtained in the transforming.

Abstract: Disclosed herein are systems, methods, and computer-readable storage devices for processing audio signals. An example system configured to practice the method receives audio at a device to be transmitted to a remote speech processing system. The system analyzes one of noise conditions, need for an enhanced speech quality, and network load to yield an analysis. Based on the analysis, the system determines to bypass user-defined options for enhancing audio for speech processing. Then, based on the analysis, the system can modify an audio transmission parameter used to transmit the audio from the device to the remote speech processing system. The audio transmission parameter can be one of an amount of coding, a chosen codec, an amount of coding, or a number of audio channels, for example.

Abstract: An audio processing device includes a first microphone configured to receive a first signal a second microphone configured to receive a second signal, a noise reduction gain determination circuit configured to determine a noise reduction gain based on the first signal and the second signal, a noise reduction circuit configured to attenuate the first signal based on the determined noise reduction gain, and an output circuit configured to output the attenuated signal.

Abstract: An audio signal processing device includes: a short-time fast Fourier transform unit 4 that generates a signal in a frequency domain obtained by performing a short-time fast Fourier transform on an input audio signal; a steady sound determining unit 5 that determines whether a waveform of a peak portion included in a waveform of the signal in a frequency domain is a steady sound; a filter coefficient calculation unit 6 that dynamically calculates a filter coefficient on the basis of a result of determination made by the steady sound determining unit 5; a comb filter 7 that operates according to the filter coefficient calculated by the filter coefficient calculation unit 6 so as to filter a signal in a frequency domain; and an inverse Fourier transform unit 8 that transforms an output of the comb filter 7 into a signal in a time domain and outputs the signal in a time domain.

Abstract: Methods and apparatus for performing signal processing. The signal processing comprises demultiplexing input encoded data into data including information for a segment including frames and coefficient information for a coefficient selected in the frames of the segment, and low band encoded data, decoding the low band encoded data to produce a low band signal, selecting a coefficient of a frame to be processed from a plurality of the coefficients based on the data, calculating a high band sub-band power of a high band sub-band signal of each sub-band constituting a high band signal of the frame to be processed based on a low band sub-band signal of each sub-band constituting the low band signal of the frame to be processed and the selected coefficient, and producing the high band signal of the frame to be processed based on the high band sub-band power and the low band sub-band signal.

Abstract: Methods and systems for audio source separation in real-time are described. In an embodiment, the present disclosure describes reading and decoding an audio source into PCM samples, fragmenting Pulse Code Modulation (PCM) samples into fragments, transforming fragments into spectrograms, performing audio source separation using a training database that includes a training dictionary and non-negative matrix factorization (NMF) to generate a set of component signals, and streaming the component signals to a playback engine. In an embodiment, a semantic equalizer graphical user allows for fading of individual component signals.

Abstract: Systems and methods are disclosed herein for improving audio conferencing services. One aspect relates to processing audio content of a conference. A first audio signal is received from a first conference participant, and a start and an end of a first utterance by the first conference participant are detected from the first audio signal. A second audio signal is received from a second conference participant, and a start and an end of a second utterance by the second conference participant is detected from the second audio signal. The second conference participant is provided with at least a portion of the first utterance, wherein at least one of start time, start point, and duration is determined based at least in part on the start, end, or both, of the second utterance.

Abstract: A tape head is provided for use with a tape drive that is configured to receive a length of tape. The tape head includes a head body including at least one head element for performing read and/or write operations on the tape, and a protective layer extending over at least a portion of the head body for inhibiting wear of the head body when the tape is moved with respect to the head body. Furthermore, the protective layer is made of titanium oxide, chromium oxide, zirconium oxide, aluminum oxide, or zinc oxide.

Abstract: Method and apparatus for enhancing write current switching efficiencies during data write operations in a data storage device. In some embodiments, write data are described in the form a sequence of symbols of nT length where T is a channel clock rate and n is an integer over a selected range. Bi-directional write currents are applied to a write element to record the sequence of symbols to a storage medium. The write currents are switched between a first rail current and a second rail current for alternating symbols. The write currents are further transitioned to an intermediate current value for at least one channel clock period immediately preceding a next occurrence of a symbol boundary between an adjacent pair of symbols in the sequence.

Abstract: A tape head is provided for reading and/or writing to a magnetic tape. The tape head including a step-like cross-sectional profile, so as to exhibit a riser between two treads, the latter respectively formed by a tape-bearing surface and a recessed surface, wherein: the tape-bearing surface is essentially flat and configured to contact a magnetic tape, and comprises at least one transducer, the latter being a read or a write element, configured to read or write to the magnetic tape, respectively; and the recessed surface is recessed from the tape-bearing surface by a distance h corresponding to a height of the riser, a width w of the recessed surface along a direction parallel to a longitudinal direction z of circulation of the tape being such that a ratio h/w is at least of 0.01. Related tape head apparatuses for recording or reproducing multi-track tapes are also provided.

Abstract: A tape head including a body with a tape-bearing surface configured to contact a magnetic tape, at least one transducer that is a read or write element, configured so the tape head may read from or write to the tape, in operation; and a monobloc closure with a structured cross-sectional profile, so as to exhibit: contact part, fixed on a side of the body which adjoins the tape-bearing surface at an edge thereof, the contact part having a top surface level with the tape-bearing surface; and connecting part integral with the contact part, the connecting part having a top surface recessed from the contact part's top surface, perpendicularly to a contact plane defined by said tape-bearing surface, so the connecting part's top surface does not contact the tape, in operation; and a broken line of mechanical weakness that extends at an end of said third surface.

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: A magneto-resistive effect element (MR element) has an upper shield that is magnetized in a cross track direction, a lower shield that is positioned at an interval relative to the upper shield in a down track direction, and a multilayer film that is positioned between the upper shield and the lower shield and that faces an air bearing surface (ABS). The multilayer film has a free layer where its magnetization direction fluctuates relative to an external magnetic field, a pinned layer where its magnetization direction is pinned against the external magnetic field, a nonmagnetic spacer layer that is positioned between the free layer and the pinned layer, and an insulating layer that is positioned at a back side of the free layer viewed from the ABS. The MR element further has a pair of side shields that are positioned at both sides of the free layer and the insulating layer in a cross track direction.

Abstract: A tape head including a body exhibiting a tape-bearing area is provided. The body comprises at least one transducer that is a read element or a write element, configured in the tape head so as for the tape head to read from or write to a magnetic tape, in operation. The tape-bearing area is essentially covered by an electrically conducting layer of material. This way, the exposed surface of the electrically conducting layer essentially forms the tape-bearing surface of the tape head, which surface contacts the magnetic tape, in operation. A tape head apparatus for recording or reproducing multi-track tapes including the tape head is also provided.

Abstract: A flexure mounted on a load beam of a disk drive suspension, including a metal base formed of a stainless-steel plate; and a conductive circuit arranged along the metal base. The conductive circuit includes an insulating layer on the metal base; and a conductor on the insulating layer. The conductor includes a thin conductor portion formed in a first region (A1) which is part of the conductive circuit in a longitudinal direction; and a thick conductor portion formed in a second region (A2) which is another part of the conductive circuit in the longitudinal direction, the thick conductor portion being thicker than the thin conductor portion. The flexure includes a flexure tail portion having an opening formed in the metal base; a bent portion of the metal base which is bent in a thickness direction at the opening; and a trace curved portion curved in the thickness direction at a position facing the opening. The trace curved portion includes the thick conductor portion.

Abstract: A storage device includes a disk, a head configured to carry out data writing and data reading with respect to the disk, and including a heater that generates heat to cause the head to thermally expand towards the disk, a non-volatile semiconductor memory, and a controller configured to set an amount of expansion of the head, based on data read from the disk during a predetermined period of time after the storage device has been turned on, and write data from a host in the disk or the non-volatile semiconductor memory during the predetermined period of time, based on the amount of expansion.

Abstract: The invention generally relates to a system for achieving reduced head-media spacing (HMS) in a hard disk drive (HDD) by providing active control over the positioning of a read/write head relative to a rotating magnetic data recording and reading surface of a disk. The system is configured to monitor and adjust positioning of the read/write head in real, or near-real time during disk rotation to maintain the HMS between a magnetic transducer of the read/write head and the disk surface within a defined range, generally between 1.0 nm and 10.0 nm, and, in some embodiments, approximately 4.3 nm. The system replaces the conventional an air bearing surface (ABS) design with a fully active suspension to maintain the head flying height within the desired range. The system further allows for each of the magnetic transducer and disk surface to be to be devoid of any overcoat or lubricant layers.

Abstract: An optical information recording/reproducing device which reproduces information from an optical information storage medium in which an interference pattern between a signal beam and a reference beam is recorded as page data includes a controller, a laser light source, a split part, an angle variable part, an optical part, a first light detector, and a second light detector, and the controller controls the angle variable part based on a control signal generated from a first signal detected by the first light detector and a second signal detected by the second light detector.

Abstract: A disc cassette includes a curved portion configured to hold multiple discs. A platform portion abuts the curved portion. A movable bale is configured to maintain discs securely in the disc cassette.