Abstract: Methods and systems for processing inbound fax messages in an efficient and scalable manner are provided. According to one embodiment, an inbound fax call is received by an Internet fax system. The fax call has a source address, a destination address and a fax signal. The Internet fax system is coupled to a circuit-switched network and a packet-switched network and includes multiple fax processing resources having different configurations, features, capabilities or capacities. One or more desired fax protocol technical call processing capabilities are determined for processing the fax signal based on one or more of the source address, the destination address, the type of network through which the fax call was received and a service provider through which the fax call was received. Then, a fax processing resource that has the desired fax protocol technical processing capabilities is dynamically selected to translate the fax signal into a digital representation.

Abstract: A method for processing a non-normalized local exchange routing guide (LERG) table to produce a normalized LERG table. The LERG tables have an NPA field, an NXX field, a Block_Id field, and a Switch_Id field. The method includes joining all distinct combinations of NPA and NXX in the non-normalized LERG table to all valid Block Ids to produce a NormLERG table. The method further includes populating a Switch Id field, in each NormLERG record having a distinct NPA-NXX value. The populating of the Switch_Id field is based on a number of corresponding non-normalized LERG records and on the Switch_Id field in one of the corresponding non-normalized LERG records and on a number of distinct Switch ID values in the corresponding non-normalized LERG records. Each of the corresponding non-normalized LERG records has a same one of the NPA-NXX values.

Abstract: Methods for adding minutes to a calling plan having a calling plan period. The methods including determining whether a threshold number of minutes have been reached during the calling plan period; and incrementally adding minutes to the calling plan when the threshold number of minutes has been reached. The cost per minute of the added minutes is less than the high-cost per minute rate normally associated with exceeding the threshold number of minutes during the calling plan period.

Abstract: A communications system includes both cellular and land based telecommunications such as PSTN telephone, as well as cellular telephone. A subscriber may use prepaid cellular phone to make and receive calls via cellular network. A Mobile Switching Center (MSC) routes the subscriber's calls to and/or from wireline network for domestic connections and/or network for international connection. A Mobile Virtual Network Operator (MVNO) operates a prepaid vendor service control point (SCP), Prepaid Vendor Service Platform (PSP), and connected databases. A method of providing a universal communications account includes: providing at least one account to a customer, enabling use of the account for a first type of connection and a second type of connection; and debiting the account based at least in part on one of the first type of connection and the second type of connection.

Abstract: A system and method for optimizing the price of a call to a business and the placement of a business' information within an ad buy on a Web page are disclosed. A call is received by a client from a potential consumer of a product. The call is transcribed into a text call stream on both the consumer and the client sides of the call. Patterns are extracted from the call. The patterns extracted may be applied against patterns relevant to the client. Patterns relevant to the client may be tied to certain goals of the business of the client that indicate successful business transactions between the consumer and the client or that indicate a service provide by the client. Extracted patterns may be used for targeting of advertisements based on a call containing patterns relevant to the client.

Abstract: Disclosed is a method and apparatus for providing telecommunication services based on customer groups. A customer group is offered a service plan that provides an incentive to early subscribers in the form of a more attractive pricing plan (i.e., discounts). For example, a particular subscriber may be provided a discount for each customer in the group who becomes a subscriber after the particular subscriber. The discounts and pricing plan may also be based, at least in part, on the total number of customers in the group that become subscribers. Further, in order to reduce customer churn, customer discounts or other incentives may be applied toward the end of a subscription plan period. Further, subscriber discounts and/or credits may be banked in a subscriber account.

Abstract: An Internet protocol query for Internet protocol private branch exchanges including a private branch exchange that is connected to the public switched telephone network for receiving a call setup request, the private branch exchange not being enabled for receiving calling name delivery information from a telecommunications circuit; at least one of an incumbent local exchange carrier switch and a competitive local exchange carrier switch in communication with the private branch exchange, the at least one of the incumbent local exchange carrier switch and the competitive local exchange carrier switch having the telecommunications circuit not enabled for calling name delivery information retrieval; a network in communication with the private branch exchange; and a calling name delivery database in communication with the network for receiving calling name delivery queries from the private branch exchange based on the call setup request. Methods for routing calls based on ENUM information is also included.

Abstract: A method and system for conference call scheduling using e-mail. The method and system can receive and parse an e-mail received from a conference moderator, containing scheduling and participant information and schedule a conference call with minimal further user intervention. In essence, the method and system can operate as a virtual administrative assistant to identify and collect telephone contact information for each of the proposed conference call participants, using a customizable hierarchy of internal and external sources of contact information. In this manner, the conference moderator does not need know telephone contact information for each conference call participant.

Abstract: A system is configured to receive an indication that a connection with a user device is to be lost, the connection being associated with a call between the user device and another user device; provide an instruction, to one or more devices, to maintain the call when the connection is lost, the one or more devices being responsible for processing the call; detect that the connection is lost; provide, to the one or more devices, a first notification that the connection is lost, the first notification causing the one or more devices to maintain the call based on the instruction to maintain the call when the connection is lost; detect that the connection is re-established; and provide, to the one or more devices, a second notification that the connection is re-established, the second notification causing the one or more devices to re-establish the call.

Abstract: A method, system and apparatus for implementing a Multimedia Ring Back Tone (MRBT) service are provided. The method includes: receiving a call request originated by a calling terminal; parsing a tone playing policy specifying whether to play a caller tone or a callee tone or whether to filter a tone; performing caller tone media negotiation or callee tone media negotiation, or performing no tone negotiation according to the parsing result; and playing the caller tone or the callee tone to the caller, or playing no tone. With the technical solution of the present invention, the caller tone service can be implemented in the IMS domain. Whether a caller tone or a callee tone is played to the caller is determined according to a preset policy. Furthermore, the tone filtering service is also implemented. This gives a user freedom in experiencing the MRBT service.

Abstract: In a click-to-call (C2C) communication environment, a web client of a computing terminal is configured to initiate a C2C call when a user selects a call tag provided in a web page being presented to the user. The call is intended to be established between a remote endpoint associated with the call tag and one of either the computing terminal or another terminal associated with the user. If the C2C call is directed to an automated system and placed in a queue or on hold, the present invention allows supplemental content intended for the user to be delivered to the web client from which the C2C call was initiated. The web client may present the supplemental content to the user.

Abstract: One or more aspects of the disclosure relate to various functions and processes related to routing of calls originating from other service providers. For example, a method includes receiving a request to route a call or a call session, such as a session initiation request, from a second service provider and proceeding to route the requested call or session through a call routing facility of the first service provider to a requested destination of the call. Routing the call through the call routing facility may include determining a service provider type and/or an origination number type for the call, determining routing information for the call using the type information, and causing routing of the call to a destination of the call based on the routing information. Further aspects of the disclosure relate to various functions and processes related to accessing routing data and/or information from a routing database.

Abstract: A contact center is described along with various methods and mechanisms for administering the same. The contact center proposed herein provides the ability to, among other things, support deferring work item routing decisions for a predetermined amount of time even when agents that are technically qualified to handle the work item are available. The deferment of work item routing decisions helps to achieve better matching and, therefore, increases contact center efficiency even though decisions are delayed.

Abstract: Methods and apparatuses for improved echo cancellation are disclosed. In one example, an audio signal to be output at a loudspeaker is received and processed to generate an optimized audio signal. The optimized audio signal produces a reduced distortion when output at the loudspeaker. The method further includes utilizing the optimized audio signal to reduce echo in a transmit audio signal.

Abstract: Methods, apparatuses, computer program products, devices and systems are described that carry out receiving level-one encrypted data including at least one associated encrypted identifier; encrypting with a level-two encryption key at least a part of the level-one encrypted data to produce level-two encrypted data; receiving a hash of the at least one associated encrypted identifier; associating the hash with the level-two encrypted data; and transmitting the level-two encrypted data and associated hash of the at least one associated encrypted identifier.

Abstract: A parallelizable integrity-aware encryption technique is provided. In at least one embodiment of the present disclosure, a parallelizable integrity-aware encryption method comprises whitening at least one message block with a first mask value, encrypting the whitened at least one message block using a block cipher and a first key, and whitening the encrypted at least one message block with a second mask value to generate at least one corresponding output ciphertext block. In another embodiment of the present disclosure, a parallelizable integrity-aware encryption method comprises applying a XOR function to all blocks of a message to compute a XOR-sum, applying a first mask value to the XOR-sum; encrypting the masked XOR-sum using a block cipher and a first key, and applying a second mask value to the encrypted XOR-sum to generate an integrity tag.

Abstract: In a wildcard-applicable anonymous hierarchical identity-based encryption system, it is aimed to make the number of pairing operations a fixed number. A user identifier ID and a pattern P are used. The user identifier ID includes k number of hierarchy identifiers. The pattern P includes either of a hierarchy identifier, a wildcard value and a blank value for each hierarchy. An encryption device 400 extracts a key value H corresponding to a hierarchy of the wildcard value and a key value H corresponding to a hierarchy of a pattern value from a public key PK, calculates a cipher value C1 and a cipher value C3, and outputs ciphertext data CT including plaintext cipher value C0, the cipher value C1, and the cipher value C3.

Abstract: Methods, systems, and computer programs for producing hash values are disclosed. A first hash value is obtained by applying a first hash function to a first input. The first input can be based on an implicit certificate, a message to be signed, a message to be verified, or other suitable information. A second hash value is obtained by applying a second hash function to a second input. The second input is based on the first hash value. The second hash value is used in a cryptographic scheme. In some instances, a public key or a private key is generated based on the second hash value. In some instances, a digital signature is generated based on the second hash value, or a digital signature is verified based on the second hash value, as appropriate.

Abstract: A system includes a build system processor configured to generate a private encryption key and configured to add the private encryption key to at least one of a plurality of source files. Each of the plurality of source files includes instructions configured to be executed by an industrial controller processor. The processor is also configured to generate a plurality of binary files from the plurality of source files. The processor is further configured to construct a whitelist file including first hash key values determined for the plurality of binary files and configured to encrypt the whitelist file using the private key to provide an encrypted whitelist file.

Abstract: A constant multiplier inputs a base and a modulo n, performs modular exponentiation modulo n with a prescribed constant as the exponent and with base a, and outputs the result of this calculation as base b. A personal key converter inputs a personal key d and calculates a personal key d? as the quotient when d is divided by the prescribed constant. A correction key generator generates a correction key d? as the remainder of the aforementioned division. A first modular exponentiation unit performs modular exponentiation base b with d? as the exponent. A second modular exponentiation unit performs modular exponentiation base a with d? as the exponent, and outputs a correction value. A correction calculation unit multiplies the outputs of the first and second modular exponentiation units and outputs the result as the encryption processing result.

Abstract: An access terminal apparatus provides a group determination key being a decryption key to a data center apparatus previously, and then transmits grouping information generated from a keyword of storage target data and having been encrypted to the data center apparatus, with encrypted data of the storage target data and tag data of the encrypted data of the storage target data. The data center apparatus stores the encrypted data relating it to the tag data, decrypts a part of the grouping information with the group determination key, derives an index value from a bit value obtained by the decryption, and stores the derived index value and the tag data relating them with each other.

Abstract: A system for accessing a service comprises two contact-less readers and one portable device. A first contact-less reader is configured to send to the portable device, through a first range radiofrequency link, a first message comprising first data. The first data includes an identifier allocated to the portable device and an associated key for encrypting data. The portable device is configured to send to a second contact-less reader, through a second range radiofrequency link, a second message comprising second data. The second data includes the identifier received from the first message. The second data is encrypted by using the data encryption key received from the first message. The second range radio frequency is higher than the first range radio frequency.

Abstract: Methods and apparatuses provide personalized radio by outputting selected songs from multiple channels from one or more source streams. Users can specify favorite channels for building their personal playlists, or multiple default playlist channels can be provided by genre or other criteria. User like/dislike inputs via the radio receiver generate filter data used to search all channels for matches or block songs from playback. Filter data can also be based on virtual DJ recommendations. Channel searching for matches involves monitoring real-time song and artist labels or other metadata for all channels carried in a separate data channel. Personalization can also be provided by using filter data to select content from a mixed channel comprising content from a selected group of channels. Connectivity options and web interfaces are provided to facilitate transfer and sharing of customization parameters for personalized radio configuration.

Abstract: An electronic device with a speaker, an amplifier for driving the speaker and a speaker grill provided on the front surface of the speaker includes a pickup detection unit for detecting that the electronic device has been picked up from water and a water discharge sound generation unit for generating a water discharge sound signal for discharging water having entered inside the speaker grill according to the detection of the pickup from the water and inputting the generated water discharge sound signal to the amplifier.

Abstract: Techniques for scene change detection around seed points in media data are provided. Media features of many different types may be extracted from the media data. One or more statistical patterns of media features in a plurality of time-wise intervals around a plurality of seed time points of the media data may be determined using one or more types of features extractable from the media data. At least one of the one or more types of features comprises a type of features that captures structural properties, tonality including harmony and melody, timbre, rhythm, loudness, stereo mix, or a quantity of sound sources as related to the media data. A plurality of beginning scene change points and a plurality of ending scene change points in the media data may be detected, based on the one or more statistical patterns, for the plurality of seed time points in the media data.

Abstract: An apparatus and method for outputting audio signals of a portable communication device are provided. The apparatus includes a plurality of audio tables including different volume control values or tone control values, and allows a user to selectively set audio tables corresponding to each of audio paths equipped with the portable communication device. Further, the apparatus analyzes a surrounding environment during a voice call, determines the absence or presence of noise, and suitably controls a volume or a tone of audio signals based on the result of the determination before outputting the audio signals. Accordingly, the apparatus can provide an improved calling condition in various calling environments.

Abstract: A harmonic echo power estimator estimates power of echo generated by harmonic loudspeaker nonlinearities in a user equipment having an echo path between a loudspeaker input and a microphone output. The estimator includes a frequency band mapper that maps each frequency band in a set of loudspeaker output signal frequency bands into a corresponding array of loudspeaker input signal frequency bands, where each frequency band in the set is mapped into several frequency bands in the corresponding array. A power estimator determines a power estimate of each input signal in each array of frequency bands. A power estimate combiner combines determined power estimates in each array of frequency bands into a corresponding estimate of loudspeaker input power generating harmonic loudspeaker nonlinearities. A power estimate transformer transforms the estimates of loudspeaker input power across the echo path into power estimates of the echo generated by the harmonic loudspeaker nonlinearities.

Abstract: The disclosure is directed to techniques for noise handling in audio or video recording. Sounds that may be generated during the audio recording, such as by activation of switches, may appear as undesired artifacts in the recording. Techniques and apparatus are described whereby, when an undesired sound is generated, an inverse of the sound is applied to the sound, thereby canceling the unwanted sound from the recording. Further, the techniques can be adapted to many kinds of portable electronic devices having many kinds of switches and many kinds of undesirable sounds.

Abstract: A personal audio device, such as a headphone, includes an adaptive noise canceling (ANC) circuit that adaptively generates an anti-noise signal from a reference microphone signal that measures the ambient audio, and the anti-noise signal is combined with source audio to provide an output for a speaker. The anti-noise signal causes cancellation of ambient audio sounds that appear at the reference microphone. A processing circuit uses the reference microphone to generate the anti-noise signal, which can be generated by an adaptive filter. The processing circuit also models an acoustic leakage path from the transducer to the reference microphone and removes elements of the source audio appearing at the reference microphone signal due to the acoustic output of the speaker. Another adaptive filter can be used to model the acoustic leakage path.

Abstract: A personal audio device including earspeakers, includes an adaptive noise canceling (ANC) circuit that adaptively generates an anti-noise signal for each earspeaker from at least one microphone signal that measures the ambient audio, and the anti-noise signals are combined with source audio to provide outputs for the earspeakers. The anti-noise signals cause cancellation of ambient audio sounds at the respective earspeakers. A processing circuit uses the microphone signal(s) to generate the anti-noise signals, which can be generated by adaptive filters. The processing circuit controls adaptation of the adaptive filters such that when an event requiring action on the adaptation of one of the adaptive filters is detected, action is taken on the other one of the adaptive filters. Another feature of the ANC system uses microphone signals provided at both of the earspeakers to perform processing on a voice microphone signal that receives speech of the user.

Abstract: A system and method for reducing or entirely canceling background or ambient noise from a voice transmission from a communications device. A communications device, such as a mobile telephone, is configured with an ambient noise compensation signal generator that is connected between a microphone and a mixer. The original output of the microphone and a compensation signal generated by the ambient noise compensation signal generator are mixed together prior to being passed to a transmitter. In one embodiment a buffer is provided between the microphone and the mixer to help synchronize the timing of the signals to be mixed. In another embodiment a second microphone is employed to detect ambient noise.

Abstract: An earphone jack circuit includes an earphone jack configured to receive an earphone plug, a frequency modulation (FM) signal processing circuit electrically connected to the earphone jack, and an amplifying circuit. The FM signal processing circuit includes a first matching circuit electrically connected to the earphone jack, and an amplifying circuit electrically connected to the first matching circuit. The first matching circuit receives an initial FM signal from the earphone jack and executes a first filtering process for the initial FM signal when the earphone plug is inserted into the earphone jack. The amplifying circuit amplifies the filtered FM signal.

Abstract: To avoid undesirable electrostatic discharge events while maintaining low leakage currents, earbuds may be provided with controlled electrostatic discharge paths. The discharge paths may include discrete components such as resistors or more distributed resistive components such as resistive elastomers. A resistive elastomer may be incorporated into an interior portion of an earbud between an earbud housing structure and a ground path. A resistive elastomer may also be used in forming an ear bud tip.

Abstract: The present invention provides a unidirectional microphone by adding an output from an omnidirectional condenser microphone unit and an output from a bi-directional ribbon microphone unit together. A condenser microphone unit 10 and a ribbon microphone unit 20 are connected in series via a step-up transformer 30, and the respective sound signals from the microphone units 10 and 20 are added together and are output from a source S of an FET 14.

Abstract: A microphone array includes a left microphone, a right microphone and a processor to receive a right microphone signal from the right microphone and a left microphone signal from the left microphone. The processor determines a timing difference between the left microphone signal and the right microphone signal. The processor determines whether the timing difference is within a time threshold. The processor time shifts one of the left microphone signal and the right microphone signal based on the timing difference. The processor also sums the shifted microphone signal and the other microphone signal to form an output signal.

Abstract: A sound enhancement technique that uses transfer functions ai,g of sounds that come from each of one or more positions/directions that are assumed to be sound sources arriving at each microphone to obtain a filter for a position that is a target of sound enhancement, where i denotes a direction and g denotes a distance for identifying each of the positions. Each of the transfer functions ai,g is represented by sum of a transmission characteristic of a direct sound that directly arrives from the position determined by the direction i and the distance g and a transmission characteristic of one or more reflected sounds produced by reflection of the direct sound off an reflective object. A filter that corresponds to the position that is the target of sound enhancement is applied to frequency-domain signals transformed from M picked-up sounds picked up with M microphones to obtain a frequency-domain output signal.

Abstract: A method for signal processing, receiving a time domain signal having a sample-rate Fs and generating N time domain signal bands, each having a bandwidth equal to Fs/N. Receiving the N signal bands and transforming a first time domain signal band to a frequency domain at a first resolution and a second time domain signal band to the frequency domain at a second resolution, where the first resolution may be different from the second resolution. Determining one or more first filter coefficients using the frequency domain components from the first signal band and one or more second filter coefficients using the frequency domain components from the second signal band. Transforming the first and second filter coefficients from the frequency domain to a time domain. Applying the first and second time domain filter coefficients to the first and second time domain signals, respectively.

Abstract: An improved method and arrangement for altering musical dynamics of a sound S included in a sound signal is disclosed. The altering of the musical dynamics is performed by filtering and amplification of the sound signal. The filtering is performed by the use of a parametric equalizer, the parametric equalizer having a first gain G1 and a resonance frequency f, being related to a pitch frequency fp of said sound S. The amplification is performed by an amplifier amplifying the sound signal with a second gain G2, the second gain G2 being dependent on the first gain G1.

Abstract: Techniques for optimizing a player based on the addition of a second player are disclosed. In an embodiment, when a first player no longer needs to play certain audio frequencies due to the addition of a second player, the gain of the first player is automatically increased as part of the setup process. In another embodiment, when a first player needs to play certain audio frequencies, for example due to the removal of a second player, the gain of the first player is automatically decreased. Many other embodiments are disclosed.

Abstract: A processing method and a processing apparatus for processing an output audio signal from an audio capture device. The output audio signal can be processed by a first processing module in a manner such that a preliminary signal and a first stage processed signal can be derived. The processing method includes providing power estimation signals, providing at least one cross power estimation signal, providing a leakage approximation and providing a control signal estimation.

Abstract: An apparatus comprises a volume-control user interface, a transmitter, an audio amplifier (having a corresponding amplification range), and a control circuit operably coupled to the foregoing components. This control circuit is configured to respond to a volume change made via the volume-control user interface that is within the amplification range of the audio amplifier by making a corresponding change to a volume level for an audio signal being transmitted by the transmitter. The control circuit is also configured to respond to a volume change made via the volume-control user interface that exceeds the amplification range of the audio amplifier by transmitting a volume-increase command.

Abstract: In accordance with an embodiment, a method of operating a charge pump includes providing a first programmable voltage to a plurality of clock generators having outputs coupled to first nodes of corresponding groups of charge pump capacitors, and selecting a second node of one capacitor from one of the corresponding groups of charge pump capacitors. The clock generators produce a plurality of clock signals having amplitudes proportional to the first programmable voltage.

Abstract: A piezoelectric sound transducer using a piezoelectric plastic material is developed from a substrate layer and a layer of a piezoelectric plastic material deposited thereon, the piezoelectric plastic layer covering the substrate layer not completely but having openings.

Abstract: An acoustic actuator according to an exemplary embodiment of the present invention includes: an actuator element generating a corresponding sound in response to an applied electrical signal; a supporting member supporting the actuator element so as to form a movement axis of the actuator element; and an edge member connected with the actuator element. The actuator element includes: a piezoelectric unit having at least one piezoelectric member of which a length is larger than a width by at least ?{square root over (2)} times; and a membrane unit including at least one membrane member generating the corresponding sound in response to the applied electrical signal, by being fixedly adhered to the piezoelectric unit so as to extend in a length direction of the at least one piezoelectric member and by forming wave movement in the extending direction in response to a current applied to the piezoelectric unit.

Abstract: A ceiling speaker mountable in a ceiling tile for mounting in a suspended ceiling of a room, includes a central woofer, and at least four surrounding pivoting tweeters that are independently adjustable so as to provide broad flat coverage throughout the room. A crossover network directs low frequency signals to the woofer and high frequency signals to the tweeters and an optional stereo separator circuit directs left and right signals to different pairs of the tweeters. A method for adjusting sound dispersion produced by the ceiling speaker is also disclosed.

Abstract: A head-related transfer function (HRTF) convolution method arranged, when an audio signal is reproduced acoustically by an electro-acoustic conversion unit disposed in a nearby position of both ears of a listener, to convolute an HRTF into the audio signal, which allows the listener to listen to the audio signal such that a sound image is localized in a perceived virtual sound image localization position, the method including the steps of: measuring, when a sound source is disposed in the virtual sound image localization position, and a sound-collecting unit is disposed in the position of the electro-acoustic conversion unit, a direct-wave direction HRTF regarding the direction of a direct wave, and reflected-wave direction HRTFs regarding the directions of selected one or more reflected waves, from the sound source to the sound-collecting unit, separately beforehand; and convoluting the obtained direct-wave direction HRTF, and the reflected-wave direction HRTFs into the audio signal.

Abstract: A hearing aid includes a microphone for conversion of acoustic sound into an input audio signal, a signal processor for processing the input audio signal for generation of an output audio signal; and a transducer for conversion of the output audio signal into a signal to be received by a human, wherein the signal processor includes a compressor with a compressor input/output rule that is variable in response to a signal level of the input audio signal.

Abstract: A directional characteristic of a microphone facility of a hearing system is more reliably controlled. The method determines a first feature value in respect of speech in a first signal of a microphone facility assigned to a first direction and a second feature value in respect of speech in a second signal of the microphone facility assigned to a second direction. A control value is obtained from the difference of the two feature values. The directional characteristic of the microphone facility is controlled with this control value.

Abstract: The present disclosure provides an SOI analog front circuit for a medical device, characterized in that it comprises an automatic gain control loop and a 2-order-3-bit-quantization Sigma-Delta analog-to-digital converter, wherein: the automatic gain control loop is configured to implement automatic control of loop gain and output an analog signal to the 2-order-3-bit-quantization Sigma-Delta analog-to-digital converter; and the 2-order-3-bit-quantization Sigma-Delta analog-to-digital converter is configured to convert the analog signal output from the automatic gain control loop into a digital code and output the digital code to a DSP for processing. The analog front circuit for the medical hearing aid device may be implemented by integration of three comparators, an analog-digital hybrid automatic gain control loop and a 2-order-3-bit-quantization Sigma-Delta analog-to-digital converter on a single chip using SOI process with high reliability and low power consumption.

Abstract: An audio signal transmission device includes a first light source and a second light source configured to emit a first wavelength of light and a second wavelength of light, respectively. The first detector and the second detector are configured to receive the first wavelength or light and the second wavelength of light respectively. A transducer electrically coupled to the detectors is configured to vibrate at least one of an eardrum or ossicle in response to the first wavelength of light and the second wavelength of light. The first detector and second detector can be coupled to the transducer with opposite polarity, such that the transducer is configured to move with a first movement in response to the first wavelength and move with a second movement in response to the second wavelength, in which the second movement opposes the first movement.