1485 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.

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.

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.

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 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.

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.

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.

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.

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 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.