1498 results about "Speech quality" patented technology

The present invention provides novel systems and methods for improving the voice quality of wireless-to-wireless calls or wireless-to-fixed terminal calls. This disclosure describes methods and apparatuses to determine and control the quality of service objectives in a telecommunication network involving at least one wireless terminal.

A speech synthesizing system producing a speech of an improved quality of voice by selecting a combination of speech segment most suitable for a synthesis speech unit sequence. The speech synthesizing system comprises a speech segment storage section where speech segment is stored, a speech segment selection information storage section where speech segment selection information including combinations of speech segment constituted of speech segment stored in the speech segment storage section for an arbitrary speech unit sequence and the appropriateness information representing the appropriatenesses of the combinations are stored, a speech segment selecting section for selecting a combination of speech segment most suitable for a synthesis parameter according to the speech segment selection information stored in the speech segment storage section, and a waveform generating section for generating speech waveform data from the combination of speech segment selected by the speech segment selecting section.

In conjunction with the dial-up public telephone system, voice quality lines carry videophone signals for monitoring a multitude of locations from at least one central station for use in a variety of applications, such as for security, surveillance, quality control and inspection, regulation of food and / or other standards in food-related and other facilities, market research, remote monitoring of deposit and withdrawal of funds at bank vaults, grocery chains, convenience stores, and the like. At the central station, telephonic interface apparatus is actuated by a control unit to selectively accomplish a telephonic connection with a remote location. Displays include the scene at the remote location and related graphic data. During routine operation, remote locations are displayed in sequence to at least one operator. Such operation may be interrupted either from a remote location or central station in the event of an urgency. In such an event, the concerned scrutiny location is displayed for further communication involving status, camera selection, camera configuration, audio, video and so on. Incoming calls from scrutiny stations imply an urgent situation for immediate display. The called number, by dialed number identification system (DNIS) may indicate the status while the calling number identifies the location by automatic number identification (ANI) signals. Alternative signalling and code techniques afford flexible operation as with respect to various graphic and status indications available for displays.

A text edit apparatus which presents, based on language analysis information regarding a text, a portion of the text where voice quality may change when the text is read aloud has advantages of predicting likelihood of the voice quality change and judging whether or not the voice quality change will occur. The apparatus includes: a voice quality change estimation unit (103) which estimates the likelihood of the voice quality change which occurs when the text is read aloud, for each predetermined unit which is an input symbol sequence of the text including at least one phonologic sequence, based on language analysis information which is a symbol sequence of a result of language analysis including a phonologic sequence corresponding to the text; a voice quality change portion judgment unit (105) which locates a portion of the text where the voice quality change is likely to occur, based on the language analysis information and a result of the estimation performed by the voice quality change estimation unit (103); and a display unit (108) which presents the user the portion which is located by the voice quality change portion judgment unit (105) as where the voice quality change is likely to occur.

The present invention pertains to improved communication quality and security of transmission in cellular communication networks. A customer has the option to pay different fees for different tiers of service relating to voice quality, bandwidth access, and different tiers of service relating to communications security. Higher tiers may guarantee a specific vocoder or bit rate is used, or guarantee a specific encryption protocol is use to ensure secure communications. Different tiers may be associated with customers' records for billing purposes. The network may afford high end devices higher voice quality and / or security via a lookup table indicating what level of service is associated with a given device. Calling or receiving devices may negotiate with each other to change to a more robust vocoder or bit rate to ensure a higher quality and / or security. Furthermore, the user may opt to change the quality and / or security level before or during a call.

A voice over Internet protocol (VoIP) system and more particularly, an improved VoIP system serving a distributed group of teleagents. Network parameters that affect voice quality over broadband connections are rapidly detected by generating both detected and subjective alerts when problems with the quality of a IP telephone connection are detected and correcting the detected problems. Network parameters are monitored during an actual VoIP telephone call or by using synthetic traffic. If the voice quality degrades for a particular teleagent, the teleagent is removed from the pool of available agents until voice quality improves. Alternatively, the voice quality of each teleagent may be ranked according to the service provided.

A speech processing apparatus which, in the process of performing echo canceling by using a pseudo acoustic echo signal, continuously uses an impulse response used for the previous frame as an impulse response to generate the pseudo acoustic echo signal when a voice is contained in the microphone input signal, and which uses a newly updated impulse response when a voice is not contained in the microphone input signal.

Architecture that facilitates initiation of a user feedback signal during a telephone call when perceiving reduced voice quality. In response to receiving the feedback signal, the telephone network performs a number of measurements on the system and related to the user device that can be analyzed to determine the problem, and potentially resolve the problem. The feedback signal can be initiated by any number of user input means, including selection of a key or combination of keys on a cell phone keypad that generate signals which can be interpreted by the telephone network as the feedback signal, e.g., a DTMF signal.

The present invention discloses a microphone array structure able to reduce noise and improve speech quality and a method thereof. The method of the present invention comprises steps: using at least two microphone to receive at least two microphone signals each containing a noise signal and a speech signal; using FFT modules to transform the microphone signals into frequency-domain signals; calculating an included angle between a speech signal and a noise signal of the microphone signal, and selecting a phase difference estimation algorithm, a noise reduction algorithm or both to reduce noise according to the included angle; if the phase difference estimation algorithm is used, calculating phase difference of the microphone signals to obtain a time-space domain mask signal; and multiplying the mask signal and the average of the microphone signals to obtain the speech signals of the microphone signals. Thereby is eliminated noise and improve speech quality.

A system for transferring a call from one teleagent to another in response to a loss of voice quality in a voice over Internet protocol (VoIP) communication system. Network parameters that affect voice quality over broadband connections are detected manually or automatically and an alert is generated. In response to the alert, a supervisor can transfer the call in progress to a second agent so that a better communication link can be provided and voice quality restored. Such transfer can be seamless to the caller. In different embodiments, transfer, or intercept, can occur manually or automatically (e.g., without supervisor intervention).

A wrist phone with a single button includes a molded-plastic wristband and a wristband cap. A wireless transceiver module is mounted on a printed circuit board and is contained in a component cavity between the wristband and the wristband cap. A foil antenna is attached directly to the printed circuit board and extends between an outer layer and a wrist-facing layer of a flexible strap portion of the wristband. A microphone is located between the layers of the opposite flexible strap portion of the wristband at least two centimeters away from the wireless transceiver module. The single button is formed using the interleaved fingers of a landing pad on a surface beneath the inner surface of the wristband. The inner surface of the wristband is covered by a layer of conductive material that makes an electrical connection between the interleaved fingers when the outer surface of the wristband is pressed.

Disclosed is a method of connecting a telephone via an IP network. An exchange on the originating side (1) evaluates voice quality along a route through the IP network to an exchange on a terminating side when a call originate request has been issued to connect the originating telephone via the IP network, (2) connects the originating telephone to the terminating telephone via the above-mentioned route if the voice quality is good, and (3) if the voice quality is bad, controls a connection of the telephone in accordance with a command from the caller or by selecting another route automatically such as an alternative route through the IP network or another route through a network other than the IP network.

An objective, service-level specific voice call listening quality test scheme for a Voice Over IP (VOIP) network is presented. Test probes are deployed along the border of the VOIP network. Each test probe is capable of placing calls over the VOIP network to the other test probes at different levels of service and measuring call quality using an objective measurement algorithm such as PAMS or PSQM. The measurement results are collected on an ongoing basis to obtain information on the VOIP network's voice call quality. The information is compared to thresholds to measure performance against Service Level Agreement guarantees.

A source signal (e.g. a speech sample) is processed or transmitted by a speech coder 1 and converted into a reception signal (coded speech signal). The source and reception signals are separately subjected to preprocessing 2 and psychoacoustic modelling 3. This is followed by a distance calculation 4, which assesses the similarity of the signals. Lastly, an MOS calculation is carried out in order to obtain a result comparable with human evaluation. According to the invention, in order to assess the transmission quality a spectral similarity value is determined which is based on calculation of the covariance of the spectra of the source signal and reception signal and division of the covariance by the standard deviations of the two said spectra.The method makes it possible to obtain an objective assessment (speech quality prediction) while taking the human auditory process into account.

A method of reducing speech delay and improving speech quality in digital transmission systems by implementing a single coder-decoder (codec) for an entire speech path from a calling party to a called party. The method is applicable to calls between mobile subscribers and between a mobile subscriber and an H.323 client terminal in a data network. After obtaining location information for the called party, an indication of a requested codec is sent from the calling party to the called party utilizing Integrated Services Data Network User Part (ISUP) or Primary Rate Interface (PRI) out-of-band signaling during call setup. If the called party supports the requested codec, a confirmation of the requested codec is sent from the called party to the calling party. A speech path is then established between the parties, and a single codec translation is performed for the entire speech path. If the called party does not support the calling party's codec, but the serving MSCs for both parties support the calling party's codec, the calling party's codec is utilized all the way to the called party's serving MSC. One additional codec translation is then performed in the called party's serving MSC to convert to the codec supported by the called party.

The embodiment of the invention discloses a network selection method, which is applied to user identification module SIM card-based terminal equipment, wherein the SIM card is used for storing related information among a user, a first network and a second network, and the first network has low data transmission rate compared with the second network but has good communication quality under most conditions. The method comprises the following steps of: preferentially lingering a CS (circuit switching) domain in the first network, and preferentially loading CS domain service through the first network; preferentially lingering a PS (packet switching) domain in the second network, and preferentially loading PS domain service through the second network; and when the communication quality of the second network does not meet the requirement of the PS domain service, lingering the PS domain to the first network, and loading the PS domain service through the first network. The method can avoid call drop and voice quality decline in a call process at the same time of ensuring high-speed data transmission, and improves the user experience of the voice service.

A method and system for objectively evaluating the quality of speech in a voice communication system. A plurality of speech reference vectors is first obtained based on a plurality of clean speech samples. A corrupted speech signal is received and processed to determine a plurality of distortions derived from a plurality of distortion measures based on the plurality of speech reference vectors. The plurality of distortions are processed by a non-linear neural network model to generate a subjective score representing user acceptance of the corrupted speech signal. The non-linear neural network model is first trained on clean speech samples as well as corrupted speech samples through the use of backpropagation to obtain the weights and bias terms necessary to predict subjective scores from several objective measures.

Techniques to enhance voice signals in a dual microphone system are disclosed. According to one aspect of the present invention, there are at least two microphones that are positioned in a pre-configured array. Two audio signals x1(k) and x2(k) are received and coupled to an adjusting module that is provided to control the gain of each of the audio signals x1(k) and x2(k) to minimize signal differences between the two signals. A separation module is provided to receive matched audio signals x′1(k) and x′2(k) from the adjusting module. The separation module separates the audio signals x′1(k) and x′2(k) to obtain a first audio signal s(k) containing mainly the voice and a second audio signal n(k) containing mainly the noise. An adaptive filtering module is provided to eliminate the noise component in the audio signal s(k) to obtain an estimated voice signal e_s(k) with a higher S / N ratio. Furthermore, the adaptive filtering module can be also configured to suppress echo in the audio signal s(k) at same time. The voice signal e_s(k) may be further coupled to a single-channel voice enhancement module that is configured to eliminate any residual of the noise component in the voice signal e_s(k) according to the differences between the voice signal and the noise signal in time domain and frequency domain, whereby, the S / N ratio is further enhanced.

The present technology provides adaptive noise reduction of an acoustic signal using a sophisticated level of control to balance the tradeoff between speech loss distortion and noise reduction. The energy level of a noise component in a sub-band signal of the acoustic signal is reduced based on an estimated signal-to-noise ratio of the sub-band signal, and further on an estimated threshold level of speech distortion in the sub-band signal. In embodiments, the energy level of the noise component in the sub-band signal may be reduced to no less than a residual noise target level. Such a target level may be defined as a level at which the noise component ceases to be perceptible.

The invention discloses a method for detecting the conditions of microphone voice-grade channels and a terminal, and belongs to the field of mobile terminals. The method includes the steps that acoustical signals are acquired through each microphone; the characteristics of each acquired acoustical signal are extracted; whether abnormal conditions of the microphone voice-grade channels exist or not is judged according to the difference between the characteristics of the acoustical signals. According to the method, whether the abnormal conditions of the microphone voice-grade channels exist or not is judged through analysis of the difference between the characteristics of the acoustical signals collected through the microphones, signal optimization is carried out on an abnormal channel or a user is reminded to examine and handle the abnormal channel, and therefore speech quality can be improved.

Architecture that facilitates initiation of a user feedback signal during a telephone call when perceiving reduced voice quality. In response to receiving the feedback signal, the telephone network performs a number of measurements on the system and related to the user device that can be analyzed to determine the problem, and potentially resolve the problem. The feedback signal can be initiated by any number of user input means, including selection of a key or combination of keys on a cell phone keypad that generate signals which can be interpreted by the telephone network as the feedback signal, e.g., a DTMF signal.

A speech separating apparatus includes: a PARCOR calculating unit (102) that extracts vocal tract information from an input speech signal; a filter smoothing unit (103) that smoothes, in a first time constant, the vocal tract information extracted by the PARCOR calculating unit (102); an inverse filtering unit (104) that calculates a filter coefficient of a filter having a frequency amplitude response characteristic inverse to the vocal tract information smoothed by the filter smoothing unit (103), so as to filter the input speech signal using the filter having the calculated filter coefficient; and a voicing source modeling unit (105) that cuts out, from the input speech signal filtered by the inverse filtering unit (104), a waveform included in a second time constant shorter than the first time constant, so as to calculate, for each waveform that is taken, voicing source information from the each waveform.

Monitoring voice quality passively across a telecommunications network and report monitoring data to a central network management system. Network is monitored for potential voice quality issues for pro-active isolation of problems prior to customer complaints about the problems. A wide cross-section of voice quality related data for IP and other networks is gathered and correlated together to provide voice quality assessments of network performance.

A method and apparatus to dynamically control selectable vocoder rates in a CDMA network to optimally improve forward and reverse link capacity as necessary. To improve capacity, a number of vocoder rates are used, each requiring a different average data rate and providing a different level of voice quality. Vocoder rates are reduced to lower sector average output power thereby improving forward and reverse link capacity. Forward and reverse link loading are compared to established thresholds that, if exceeded, would trigger a forward link and a reverse link vocoder rate adjustment as necessary to improve network capacity. The forward link vocoder rate and reverse link vocoder rate may be set independently or the reverse link vocoder rate may be set as a function of the forward link vocoder rate to maintain a link balance.

An apparatus and a method of optimizing voice quality on a network having end-points that are voice over Internet Protocol (IP) devices. Default parameters of the end-points are initialized. Network performance parameters are measured and evaluated to determine whether they signify that connection to the network is below a desired level of operation. If so, the default parameters of the end-points are re-set based on the evaluation. The adjustment may entail re-negotiating the CODEC connection and re-setting parameters for packet size and re-setting parameters for jitter buffer size.

A system and a method are provided for evaluating a voice performance in order to recognize a long-distance voice. The system implements a voice performance evaluation function for long-distance voice input in a robot. Particularly, in robots including a network robot, it is required to normally perform voice recognition so that a speaking subject and a surrounding situation can be recognized by a robot. Accordingly, in order to obtain the most optimal voice quality, it is important to find a noise removal algorithm through an optimal hardware configuration and an optimal combination of the optimal hardware configuration and software. Therefore, a method for finding a noise removal algorithm appropriate for each of cases, including one case where a distance from a speaking subject is fixed and another case where a distance from a speaking subject changes. As a result, the most optimal voice quality can be obtained regardless of a noise environment even when the speaking subject is a long distance away from the robot.

A method and apparatus for concealing frame loss and an apparatus for transmitting and receiving a speech signal that are capable of reducing speech quality degradation caused by packet loss are provided. In the method, when loss of a current received frame occurs, a random excitation signal having the highest correlation with a periodic excitation signal (i.e., a pitch excitation signal) decoded from a previous frame received without loss is used as a noise excitation signal to recover an excitation signal of a current lost frame. Furthermore, a third, new attenuation constant (AS) is obtained by summing a first attenuation constant (NS) obtained based on the number of continuously lost frames and a second attenuation constant (PS) predicted in consideration of change in amplitude of previously received frames to adjust the amplitude of the recovered excitation signal for the current lost frame. Speech quality degradation caused by packet loss can be reduced for enhanced communication quality in a packet network environment with continuous frame loss.

The present invention provides a novel system and a method for improving the voice quality of wireless-to-wireless calls or wireless-to-fixed terminal calls, while permitting a more efficient utilization of network resources. In a first aspect, the invention provides a communication system including a plurality of base stations connected to one another through a digital network (such as TDM, frame relay or ATM). Also, a plurality of vocoder channels, separate from the individual base stations, provide a data flow path from the base stations to wireline terminals first through the digital network via a Mobile Switching Center and finally through a landline network. In a wireless-to-wireless call, the compressed audio signal travels from one base station to another without undergoing any de-compression / compression. This avoids undesirable vocoder tandeming known to degrade voice quality. If the call is identified as being of a wireless-to-wireline terminal nature, the data is directed to a vocoder channel for decompression into PCM samples. The samples are then transmitted to the fixed terminal through the landline network. The main advantage of this network architecture is that fewer vocoders are required by comparison to prior art systems, therefore fewer costs are incurred. In addition, voice quality is improved.

An objective, service-level specific voice call listening quality test scheme for a Voice Over IP (VOIP) network is presented. Test probes are deployed along the border of the VOIP network. Each test probe is capable of placing calls over the VOIP network to the other test probes at different levels of service and measuring call quality using an objective measurement algorithm such as PAMS or PSQM. The measurement results are collected on an ongoing basis to obtain information on the VOIP network's voice call quality. The information is compared to thresholds to measure performance against Service Level Agreement guarantees.

A closed caption display controller to control a display mode of a closed caption corresponding to a speech of an audio signal, the controller comprises an analysis unit to analyze the speech on a speech quality, an examination unit configured to examine a speech listening level according to a given rule based on an analysis result of the analysis unit, and a determination unit to determine a display mode according to an examination result of the examination unit.