61results about "Speech amplifier applications" patented technology

A signal processing method including the steps of: receiving a first audio signal; detecting the presence of one or more shrieks within an audible frequency range of said audio signal; creating one or more filters to selectively attenuate the respective one or more shrieks within the audible frequency range; filtering the audio signal using the one or more filters; and transmitting the filtered audio signal to an audio telephone device.

A method and apparatus normalizes the playback loudness of stored sound recordings to avoid objectionable variations in perceived loudness between different sound recordings at the same volume setting. In an exemplary processing method, a stored sound recording is processed to determine its loudness. That loudness, or some value derived from it, is then used to set the playback gain used for playing back the sound recording. Thus, for a given volume setting, the playback gain can be set lower for louder recordings, and higher for quieter recordings. In one or more exemplary embodiments, sound recordings are processed as received, or at least some time in advance of their first playback, so that a loudness-based gain compensation parameter can be calculated and stored for them. The corresponding stored gain control parameter can then be selected and used responsive to selecting a particular sound recording for playback.

A high efficiency line driver is disclosed. The line driver may be applied with equal advantage in wired and wireless communication media to amplify data signals with a minimum of power consumption. In an embodiment of the invention a line driver is disclosed which includes: at least one amplifier, a delay element, a control signal generator and a generator. The at least one amplifier includes at least one bias supply, a signal input and a signal output. The delay element accepts as an input the data signal and delays delivery of the data signal to the at least one line amplifier for amplification. The generator is responsive to a control signal to generate varying voltage levels corresponding thereto on the at least one bias supply of the at least one amplifier. The control signal generator is responsive to the input data signal to detect peaks therein and to generate the control signal corresponding thereto in advance of delivery of the data signal to the amplifier.

System to allow a conferee that has been defined as a source of nuisance to respond to an automatic decision of an MCU. If the conferee places the conference on ‘HOLD’ with hold music, the ND may identify this as a nuisance connection and send this indication to a control unit in the MCU. The control unit may mute the signal coming from this conferee and may place an Interactive Voice Response (IVR) message to the conferee. A message may inform the conferee that he has been muted and request to press a touch tone key, for instance ‘1’, if he returned from ‘HOLD,’ to press ‘3’ if using a noisy line / environment, or press ‘5’ to disable the ND algorithm. If a response is not received, the message may continue for a certain period or for the rest of the conference. If one of those keys has been pushed the system may act as follows. If the key is ‘1’ an embodiment cancels the mute situation and enables the conferee to be heard. If the key is ‘3’ the embodiment may reduce the sensitivity of the ND algorithm. If the new level is above a defined tolerance the embodiment may instruct the conferee that he is too noisy and not add the conferee's audio to the conference. If the key is ‘5’ the embodiment may disable the ND algorithm and connect the nuisance conferee without condition. Other embodiments may place a noisy conferee in push to talk (PTT) operation.

A system to allow a conferee defined as a nuisance source to respond to an automatic nuisance decision of a Multipoint Control Unit is disclosed. For example, a conferee placing a conference on hold may generate on hold music, which may be identified by a nuisance detector. The nuisance detector sends an indication of the nuisance condition to a control unit in the MCU. The control unit mutes the signal coming from this conferee and sends an Interactive Voice Response (IVR) message to the conferee. The message may inform the conferee that he has been muted and requests the conferee for example to press one of a plurality of touch tone keys depending on his situation. For instance, the conferee can select particular keys if he is returning from the hold condition, if he is experiencing a noisy line / environment and needs to reduce the sensitivity of the nuisance detector algorithm, or if he wishes to disable the nuisance detector algorithm.

Techniques for enhancing the sound quality of computer telephony systems are provided. In general, digital signals including telephony sounds are converted from the time domain to the frequency domain. If noise is detected in the frequency domain conversion of the digital signals, the noise is removed utilizing a filter. The noise detection and filtering can be performed in software so that enhanced audio quality can be achieved at a relatively low cost and with flexibility for very diverse environments.

A Voice over Internet Protocol (VoIP) system having a dynamic gain control function and a method for providing a gain dynamically using the same system wherein, in the process that Pulse Code Modulation (PCM) data are transformed into a VoIP packet or vice versa, a gain value is dynamically assigned according to the type of terminal equipment utilized and the PCM voice data can be amplified and outputted according to the assigned gain value. The system includes a private exchange having a gain table and a VoIP gateway for the receiving the gain value from the private exchange in order to amplify the PCM voice and transform the amplified PCM voice data into the VoIP packet or convert VoIP packets to PCM voice data and amplifying the PCM voice data.

A method is proposed for reducing the number of vocoders in a wireless network. In certain wireless networks mechanisms exist for bypassing the encoding and decoding of voice traffic at the base station, when both ends of the call are mobile subscribers. This is known as tandem-free operation. A method is proposed where other elements of the network instruct the base station to bypass the vocoding, even when the other end of the link is not a wireless subscriber. These elements could include a packet-switched network used for backhaul, or a terminating node such as a voicemail platform. The network element instructing the base station to bypass its vocoders will then be responsible for transcoding of the voice data. This eliminates the need for extra stages of vocoding.

An IP telephone system includes an IP telephone apparatus, a Web server and an ENUM server. The IP telephone apparatus is connected to an IP network. The Web server registers talk volume information corresponding to a telephone number assigned to the IP telephone apparatus. The ENUM server registers a NAPTR resource record in which a URI specifies link data related to the talk volume information registered in the Web server. In the IP telephone system, the IP telephone apparatus that has received a call transmits, to the ENUM server, a query for a NAPTR record corresponding to a targeted recipient phone number. The IP telephone apparatus then transmits, to the Web server, a request for the talk volume information corresponding to the targeted recipient phone number according to the returned NAPTR resource record, and adjusts the talk volume according to the returned talk volume information.

The present invention provides a mobile phone audio reminding method comprising the following steps: (1) a mobile phone saves a reminding means at an earphone mode; the reminding means comprises an earphone plus a loudspeaker reminding means; (2) when audio reminding is required, the mobile phone outputs corresponding reminding from a corresponding audio output channel according to a current reminding means. The method of the present invention allows a user to browse short messages and receive calls in time, thereby enhancing the friendliness of the use of the mobile phone.

An IP telephone system includes an IP telephone apparatus, a Web server and an ENUM server. The IP telephone apparatus is connected to an IP network. The Web server registers talk volume information corresponding to a telephone number assigned to the IP telephone apparatus. The ENUM server registers a NAPTR resource record in which a URI specifies link data related to the talk volume information registered in the Web server. In the IP telephone system, the IP telephone apparatus that has received a call transmits, to the ENUM server, a query for a NAPTR record corresponding to a targeted recipient phone number. The IP telephone apparatus then transmits, to the Web server, a request for the talk volume information corresponding to the targeted recipient phone number according to the returned NAPTR resource record, and adjusts the talk volume according to the returned talk volume information.

A method and apparatus is provided for improving at least one gain bandwidth path. At least one signal being transmitted is monitored. A gain / bandwidth control process is performed upon the monitoring of the signal. The apparatus taught by the present invention comprises: a first circuit portion capable of driving a signal onto a subscriber line; and a second circuit portion electrically coupled with the first circuit portion, wherein the second circuit portion is capable of separating a plurality of signal paths based upon at least one characteristic of the signal path for applying an appropriate gain factor upon the signal path.

A method and apparatus normalizes the playback loudness of stored sound recordings to avoid objectionable variations in perceived loudness between different sound recordings at the same volume setting. In an exemplary processing method, a stored sound recording is processed to determine its loudness. That loudness, or some value derived from it, is then used to set the playback gain used for playing back the sound recording. Thus, for a given volume setting, the playback gain can be set lower for louder recordings, and higher for quieter recordings. In one or more exemplary embodiments, sound recordings are processed as received, or at least some time in advance of their first playback, so that a loudness-based gain compensation parameter can be calculated and stored for them. The corresponding stored gain control parameter can then be selected and used responsive to selecting a particular sound recording for playback.

A DSL line conditioner achieves high performance sufficient to simultaneously support video, voice and data signals on standard telephone twisted pair lines over substantially greater distances than are currently available. The line conditioner automatically adjusts and sets the upstream and downstream preamplifier gains and attenuations according to the actual degradation imposed upon the upstream and downstream signals by the twisted pair line in which the line conditioner is used. The line conditioner achieves high performance by optimizing the signal-to-noise ratio and signal quality of DSL signals, and has a low power budget that enables it to operate by deriving power from the twisted pair lines over which it is used.

An extension repeater which interfaces between a digital service network and a local network span combines extensive diagnostic and operational functions in a single, space-efficient device. The functions include simultaneous loopback capability in both the direction of the digital service network and the local network span, signal regeneration capability in both directions, non-intrusive monitoring connections for each side of the repeater, and visual indicators of the loopback states as well as the framing type of the data received from the digital service network and the local network span. Furthermore, the extension repeater is physically sized and shaped to fit in standard wall-mount telephone equipment shelves in order to provide economical and simplified installation of the repeater at locations outside of a telephone network central switching office.

In a time-division duplex (TDD) system, a reliable initialization scheme that is applicable to an automatic gain controller (AGC) at a base station is implemented in various forms depending on the availability of certain information such as signal-to-interference ratio (SIR), spreading factors and other parameters. A more accurate estimation of the initial control word of a gain-adjustable amplifier for one or more time slots is implemented. The scheme is applicable to AGC initialization for each time slot of the TDD system, but is also applicable to other systems of transmission, without limitation.

System and method are disclosed to mitigate noise in an area monitored by a plurality of telecommunication terminals in communication with a controller, the method including: detecting an excessive noise originating in the area monitored by the plurality of telecommunication terminals, the detection performed by one or more terminals of the plurality of telecommunication terminals; identifying a location of a first terminal of the plurality of telecommunication terminals by use of the detected excessive noise, wherein the first terminal is closest to a source of the excessive noise; and providing a mitigation message to the identified first terminal.

A method for providing selective mid-call call forwarding from a mobile station associated with a subscriber to a wireless service provider is provided. In one embodiment, the method includes: a) receiving an incoming call, b) ringing the mobile station, c) receiving a mid-call call forwarding activation from the mobile station, d) retrieving a telephone number associated with the mid-call call forwarding activation from a service plan associated with the subscriber, and e) forwarding the incoming call to a telephone device associated with the retrieved telephone number. In another embodiment, the mid-call call forwarding activation is a control signal resulting from a user pressing at least one key on the mobile station. In other aspects, a method of adding and / or modifying a selective mid-call call forwarding feature to a service plan associated with the subscriber is provided.

This invention relates to a method for realizing functions of playing prompt and signal voices including compacting the needed playing prompt voice and signal voice information according to differentphonetic coding kinds of different rates to generate related phonetic frame files, then segmenting all the phonetic frame files in terms of same time intervals and identifying each assigned segment sequential number, and therefore, when the prompt and signal voices are played, the switching can be done between different phonetic frame files to ensure the playing continuity. This invention omits the phonetic coder in the present technology and need not the sources of assigned phonetic coders.

An apparatus for improving signal quality in a communications link includes apparatus for regenerating only the speech-like characteristics of signals received over the communications link so that an estimate of the original speech signal can be retransmitted. The apparatus may include a vocal tract model coupled to a synthesizer.

The invention discloses a sound enhancement method and a sound enhancement system. The method comprises the following steps: obtaining a sound signal, and converting the sound signal into a digital signal; decomposing the digital signal to obtain a plurality of intrinsic mode functions or a plurality of similar intrinsic mode functions; selectively amplifying the amplitudes of the plurality of obtained intrinsic mode functions or the plurality of similar intrinsic mode functions; integrating the selectively amplified intrinsic mode functions or similar intrinsic mode functions to obtain an integrated reconstruction signal; and converting the integrated reconstructed signal into an analog signal. Based on Hilbert-Huang transform, sound can be effectively and selectively enhanced, only high-frequency consonants in the sound are amplified instead of amplifying vowels, the method can effectively improve the sharpness of the amplified sound, and the problem that only the loudness of the sound is increased but the sharpness is not increased in an existing sound enhancement method is solved.

A call apparatus, a call method and a call system, are disclosed, in which the problem otherwise caused by fixed setting of the sound volume level of the BGM or the effect sound with respect to the call voice may be overcome. A decoded output of a decoder 17 (PCM data) is multiplied in a gain adjustment unit 18 with a gain coefficient k2, as the SE sound volume level as set by a user. A multiplication output of the gain adjustment unit 18 is sent to an adder 13. The decoded output of a decoder 20 is multiplied in a gain adjustment unit 21 with a gain coefficient k3, as the BGM sound volume level, as set by the user. A multiplication output of the gain adjustment unit 21 is sent to an adder 13. The adder 13 sums the multiplication outputs of the gain adjustment units 12, 18 and 21 and sends a sum output to an encoder 22.

A multi-stage differential amplifier and method provides a gain versus frequency warping response characteristic that equalizes a transmission line of a given length to compensate for frequency-dependent attenuation. The warping response characteristic comprises the inverse of the attenuation versus frequency characteristic of the line, and affords a substantially flat system gain response over a predetermined frequency spectrum. The individual amplifier stages employ operational amplifiers, and are designed using SPICE simulations such that the individual responses of the stages are tailored so they combine to provide the required overall warping response to compensate the line.

A method and system of managing end-to-end loss in a manner that allows phone calls to be executed across many networks at desirable end-to-end loss levels, including networks having fixed-loss stratagems, by controlling devices to adjust their local loss or gain in order to establish target end-to-end loss.

A circuit for canceling echo and sidetone in a switching system connecting first and second subscribers, comprising a first linear converter, a sidetone canceling filter, a limiter, a first PCM converter, a second linear converter, echo cancellation filter, and a second PCM converter.

A method of processing telephone voice output is applied in an electronic device with telephone function. When any one of the two communicators is a hearing-impaired user, at least one of the electronic devices will obtain the corresponding voice adjustment parameters according to the receiver identification of the hearing-impaired user. Therefore, the voice adjusting program is able to process the voice in advance based on the voice adjustment parameters to help the hearing-impaired user hear better.