162 results about "Quality voice" patented technology

The present invention provides a system and method for a reliable, low-cost, secure Internet Protocol (IP) based network that provides broadband-based voice communications as well as video and data communications. The IP network is arranged to function with the infrastructure of the Public Switched Telephone Network (PSTN), to control telephone calls in SS7 type networks and to provide the features, applications, and services of the typical SS7 networks in a voice over IP network. The present invention supports large effective call volumes, allows accommodation of a wide range of broadband-based service platforms, provides flexibility to support current and future calling feature services, and provides high quality voice transmission.

A system and method of bypassing the regulated portion of the Public Switching Telephone Network (PSTN) to establish carrier-grade voice transmissions and/or IP data communications between an Internet Calling Person having a first telephone and a first PC coupled to a first Local Service Access Provider (LSAP) and an Internet Called Party having a second telephone and a second PC coupled to a second different Local Service Access Provider (LSAP).

Certain aspects of a method and system for providing wireless communication may include a single integrated circuit (IC) that handles communication for at least a Bluetooth (BT) connection and a Wireless Local Area Network (WLAN) connection by controlling transmission of acknowledgement signals based on a measured voice quality for one or both of the BT connection and the WLAN connection. The transmission of the acknowledgement signals communicated from the mobile station to an access point may be controlled. A BT priority signal may be asserted if data to be transmitted comprises high quality voice (HV3) data. The transmission of the acknowledgement signals may be controlled by raising a priority of an asserted WLAN priority signal higher than a priority of the asserted BT priority signal. The voice data may be transmitted via high quality voice (HV3) frames over the BT connection based on the asserted WLAN priority signal.

A method and apparatus for establishing traffic priorities in an IP network connected with multiple communication sources is disclosed. Different types of packets such as voice packets and data packets are received for transmission over a single network access link. In order to assure quality voice transmissions, voice packets are given priority over data packets by limiting a transmission rate from a data input queue whenever voice packets are detected in a voice input queue. Different combinations of interleaved voice and data packets help to alleviate packet congestion while maintaining priority for voice packets. In one aspect of the invention, priority for voice packets is further accomplished by limiting the size of data packet frames. In some instances when no voice packets are awaiting transmission, data packets are transmitted at a maximum rate in order to make full use of the network access link bandwidth. The invention is particularly beneficial when implemented over cable and DSL network access links with slow upstream feed rates.

A novel protocol for an ad-hoc, peer-to-peer radio network that provides collision-free channel access with an emphasis on improving geographic reuse of the frequency spectrum. The protocol of the invention is executed on the reservation or control channel, and provides a method for allocating data transactions on the data channels. The system of the invention utilizes multiple parallel data channels that are coordinated by a single reservation channel. The transceiver of the system employs two modems to solve the channel reliability issues with multiple channel designs, where one is dedicated as a receive-only modem for gathering channel usage information on the reservation channel. High quality voice, video and data may be transmitted. The reservation channel implements a time division multiple access algorithm with dynamic slot allocation. In a distributed manner, nodes determine geographic reuse of slots based on channel quality extracted from the modem. Signal quality calculations are used to determine the likelihood of a slot reuse causing destructive interference within a node's neighborhood. Requests for slot usage are compared with the known traffic pattern and accepted or rejected by nodes within RF signal range based on the signal quality calculations.

The embodiment of the invention discloses a microphone-array speech-beam forming method, which comprises a digital-signal converting step, a frequency-region signal-obtaining step, a time-delay obtaining step, a time-delay compensating step and a weighted stacking step, wherein time-delay estimation based on phase conversion is adopted particularly in the time-delay obtaining step, thereby a beam-forming signal pointed to a target sound-source spatial position after enhancement processing is obtained. Compared with the prior art, the positioning precision and the directivity of a three-dimensional space are enhanced, the long-distance sound-picking capability in a complicated acoustics environment is greatly enhanced, high-quality voice signals are obtained, and noise and other interferences are reduced. The embodiment of the invention also discloses a speech-signal processing device and a system.

A hands-free loudspeaker system which is capable of achieving high-quality voice amplification without requiring a human speaker to move to a microphone or a microphone to be moved to a human speaker. A microphone whose input level has continued to be above a threshold value for not shorter than a predetermined time period is detected, based on input signals from dispersedly arranged microphones. An input signal from the microphone is selected and outputted to each loudspeaker at an output level or with a delay time, according to a location of the loudspeaker. A preset lowest threshold level is initially set to the threshold value, and an input level of the microphone higher than the threshold value is newly set to the same, while when the input level is lower than the threshold value, a lower value is set to the same in a step-by-step manner.

A clear, high quality voice signal with a high signal-to-noise ratio is achieved by use of an adaptive noise reduction scheme with two microphones in close proximity. The method includes the use of two omini directional microphones in a highly directional mode, and then applying an adaptive noise cancellation algorithm to reduce the noise.

A method and apparatus for a voice transcoder that converts a bitstream representing frames of data encoded according to a first voice compression standard to a bitstream representing frames of data according to a second voice compression standard using perceptual weighting that uses tuned weighting factors, such that the bitstream of a second voice compression standard to produce a higher quality decoded voice signal than a comparable tandem transcoding solution. The method includes pre-computing weighting factors for a perceptual weighting filter optimized to a specific source and destination codec pair, pre-configuring the transcoding strategies, mapping CELP parameters in the CELP parameter space according to the selected coding strategy, performing Linear Prediction analysis if specified by the transcoding strategy, perceptually weighting the speech using with tuned weighting factors, and searching for adaptive codebook and fixed-codebook parameters to obtain a quantized set of destination codec parameters.

The invention provides a method and a system for eliminating multi-channel acoustic echo of remote voice frequency interaction. The method for eliminating the multi-channel acoustic echo of the remote voice frequency interaction comprises the following steps of: obtaining input multi-channel sound source acoustic signals; carrying out acoustic separation processing for the input multi-channel sound source acoustic signals; carrying out acoustic echo elimination processing for each-channel acoustic signals which are subjected to the acoustic separation processing through a self-adapting filter; combining and carrying out acoustic combination for the each-channel acoustic signals which are subjected to the acoustic echo elimination processing; outputting the acoustic signal which is subjected to the acoustic combination. The system comprises a DSP (Digital Signal Processor), a multi-channel acoustic separation combining device and a self-adapting filter module. According to the method and the system, provided by the invention, the multi-channel acoustic echo can be eliminated with lower cost, high-quality voice communication is supplied, the method and the system which are provided by the invention can be applied to various complicated remote voice bidirectional voice interaction fields, the smearing time for processing the echo can be increased to be 500 ms, the acoustic bandwidth is in the range of 20 Hz to 20 KHz, and the echo elimination ability is increased to be larger than 100 dB.

A clear, high quality voice signal with a high signal-to-noise ratio is achieved by use of an adaptive noise reduction scheme with two microphones in close proximity. The method includes the use of two omini directional microphones in a highly directional mode, and then applying an adaptive noise cancellation algorithm to reduce the noise.

The invention discloses a voice conversion method based on adaptive Gaussian clustering under a non-parallel text condition, and belongs to the technical field of voice signal processing. The method comprises the steps: firstly carrying out the voice feature parameter alignment of non-parallel linguistic data through a method based on the combination of unit selection and sound channel length normalization; secondly carrying out the training of an adaptive Gaussian mixed model and bilinear frequency bending and amplitude adjustment, and obtaining a conversion function needed by voice conversion; finally achieving the high-quality voice conversion through the conversion function. The method overcomes the limit that the training stage requires the parallel linguistic data, achieves the voice conversion under the non-parallel text condition, is higher in adaptability and universality, employs the adaptive Gaussian mixed model to replace a conventional Gaussian mixed model, solves a problem that a Gaussian mixed model is not precise in voice feature parameter classification, combines the adaptive Gaussian mixed model with the bilinear frequency bending and amplitude adjustment, and is better in conversion personality similarity and voice quality.

The invention provides a session server, a user terminal and a method for controlling voice quality in a voice session. The session server comprises a determination module and a first processing module, wherein the determining module is used for determining whether a voice signal which is accessed to the voice session and inputted by a first user terminal is a low quality voice signal or not to obtain a determination result; and the first processing module is used for silencing the first user terminal or breaking the connection between the first user terminal and the voice session when the determination result indicates that the voice signal inputted by the first user terminal is the low quality voice signal. The session server, the user terminal and the method can effectively control the problem of low quality voice in the voice session.

The invention discloses a high-quality voice conversion method based on modeling of signal timing characteristics. The high-quality voice conversion method based on the modeling of the signal timing characteristics comprises the following steps: aiming at parallel data of a source and a target, considering modeling and tracing the timing characteristics of the source and the target, utilizing the hybrid Kalman filter, estimating structural parameters of a model under the criteria of expectation maximization, utilizing characteristic parameter set of mapping voice of the model and finally achieving a high-quality voice conversion effect. According to the high-quality voice conversion method based on the modeling of the signal timing characteristics, strong correlation between the voice signal parameters is fully utilized, a novel hybrid Kalman filter is constructed by means of a physical process that simulation parameters change with time and is used for a parameter mapping process of the voice conversion, a set of special conversion algorithm which associates the parameter of the Kalman filter with physical properties of a voice signal is designed, and therefore a conversion of personality traits of a speaker can be achieved.

A method for high quality voice communication over an IP network. The method is implemented using an IP network device. Voice communication quality of service is initiated through the transmission of a path message for the voice communication. The path message is configured for establishing a communications path through the nodes of the IP network. A reservation message is received in response to the path message. The reservation message is configured for specifying a range of voice streams for a bandwidth reservation, allowing a single reservation message to specify bandwidth for quality of service for multiple voice calls. The bandwidth reservation for the range of voice streams is implemented in accordance with the reservation message. The path message can be transmitted from an originating IP network device, such as an originating VoIP gateway, and can be generated by a first voice application executing thereon. The reservation message can be received from a terminating IP network device, such as a terminating VoIP gateway, and can be generated by a second voice application executing thereon. The reservation message includes a source port range specifying the range of voice streams for transmission. A bandwidth reservation table within the IP network device is updated in accordance with the reservation message, and bandwidth for transmission of the range of voice streams is reserved using the reservation table.

The invention provides a method for encrypting voice based on the GSM voice channel transmission, which is applicable to the field of mobile communication. Through directly modulating encrypted voice data into voice-like coding signals conforming to the acoustic code coding format in the voice-like modulation process of the GSM sending end, the method can reduce once encoding and decoding of an acoustic code device. Even in the IP network, the signal transmission speed obtained by the method can meet the requirement of high quality voice secrecy communication.

The invention provides a specific continuous voice filtering method and device of a mobile phone, and belongs to the technical field of communication methods and equipment. When a user uses the function, user voice is recorded at first, identified specific voice is recorded in the mobile phone, and a voice library of the mobile phone is built; when the mobile phone is used for communication every time, the user can make the choice that the mobile phone only identifies the specific voice. By means of the specific continuous voice filtering method and device of the mobile phone, in a noisy communication environment, communication inconvenience occurring in the environment can be overcome effectively, and it can be guaranteed that the other party of the communication can obtain high-quality voice communication; certainly, when the function is used, only the voice of the user can pass through the mobile phone, voice of other people around the user cannot be transmitted to the other party, and high applicability is obtained in the aspect of user privacy protection.

A relay interface device base on a synchronous digit transmission system relates to E/M relay interface including a core control circuit, a clock synchronous circuit a synchronous lead-in process circuit, a time-slot cross circuit, an audio interface circuit, PCM code/decode and time-slot distribution circuit, a sending circuit of M signaling and a receiving circuit of E signaling. The original E/M analog relay interface equipments can be still used in synchronous data transferring network system with this invented devices, so as to save the initial reform expenses and increase quality voice and signaling.