76results about How to "Guaranteed Voice Quality" patented technology

An apparatus and method for processing packets in a Wireless Local Area Network (WLAN) access point includes a WLAN access point cooperated with the WLAN parses the received packets, determines whether the received packets are voice packets which must be preferentially processed or signaling packets of a protocol related to a call control in order to guarantee a quality of VoIP service, gives a priority to the corresponding packets in a differentiated manner according to the determined result, and preferentially processes the packets given a high priority. Thus, it is possible to guarantee the quality of VoIP service.

This invention relates to one method to realize real-time test network transmission time lag in RTP, which comprises the following steps: sending SR / RR system time and receiving SR / RR current system time on this end recording; computing network transmission time lag through DLSR and DLRTCP on end system time. This invention determines system time to take place NTP time no matter whether to start NTP agreement without limit of SR messages.

The invention provides a residual echo inhibition method and device. The method comprises steps of: acquiring the energy of an error signal in an echo filtering process, the cross-power spectrum density of the error signal and a near-end microphone acquisition signal, the self-power spectrum density of the error signal, the energy of the near-end microphone acquisition signal, the reverberation power spectrum density of a reference signal, and the sound attenuation coefficient of an echo; computing a filter coefficient of frequency-domain Wiener filtering according to the energy of the near-end microphone acquisition signal and the energy of the error signal; computing a filter coefficient of similar Wiener filtering according to the cross-power spectrum density of the error signal and the near-end microphone acquisition signal, the self-power spectrum density of the error signal, the reverberation power spectrum density of the reference signal, and the sound attenuation coefficient of the echo; and filtering the residual echo according to the filter coefficient of frequency-domain Wiener filtering and the filter coefficient of similar Wiener filtering. The method and the device can be suitable for varied application environments and are capable of accurately filtering the residual echo and guaranteeing good voice quality.

The invention discloses a phase-dependent shared deep convolutional neural network speech enhancement method, which comprises the following steps of: performing time-frequency domain analysis on noisyspeech data and clean speech data by using short-time Fourier transform to obtain dual-channel time-frequency spectrum characteristics of the noisy speech data and the clean speech data respectively,taking the dual-channel time-frequency spectrum characteristics as training samples, building a shared deep convolutional neural network, training the shared deep convolutional neural network by using a training sample, for to-be-enhanced noisy speech data, obtaining dual-channel time-frequency spectrum features of the noisy speech data, inputting the dual-channel time-frequency spectrum featuresinto the shared deep convolutional neural model, calculating and outputting the predicted dual-channel time-frequency spectrum features, and processing the enhanced dual-channel time-frequency spectrum features by using short-time inverse Fourier transform and an overlap-add method to obtain an enhanced speech signal. The method can effectively suppress noise interference in speech signals and enhance the quality of the speech signals.

The invention discloses a wireless sensor network access control method and the system thereof for speech acquisition and transmission and belongs to the field of wireless sensor network. The method of the invention comprises the following procedures: a node captures a terminal-to-terminal path packet loss rate from the node to a convergent node, and compression ratio and redundancy rate are calculated according to the packet loss rate; a node i is required to calculate the rate of injection according to the compression ratio and the redundancy rate and sends out a participation request message containing the rate of injection to the convergent node simultaneously; the convergent node judges whether the current flow satisfies the bandwidth constraint according to the participation request message; if the current flow satisfies the bandwidth constraint and no message of objecting the node i to participate sent by other nodes is received, then the node i is allowed to participate; if objects, the node i is not allowed to participate. The system of the invention also comprises a microcontroller, a wireless transmit-receive module and a speech acquisition device. Compared with the present technique, the invention has the advantages of ensuring the terminal-to-terminal speech quality, raising network utilization rate and providing on-line and in-real evaluation of sensor network speech transmission.

The invention discloses a voice signal bandwidth expansion method and a device thereof. The method comprises the following steps: (1) carrying out sampling rate conversion processing on a voice signal and obtaining a high frequency signal and a low frequency signal after filtering, (2) estimating high frequency signal energy and obtaining a high frequency signal energy value, (3) when the high frequency signal energy value is larger than a preset energy threshold, judging the voice signal as a broadband voice signal, carrying out gain or attenuation adjustment on the high frequency signal, and overlapping the low frequency signal with the adjusted high frequency signal for outputting, (4) when the high frequency signal energy value is smaller than or equal to a preset energy threshold, judging the voice signal as a narrowband voice signal, carrying out bandwidth expansion on the low frequency signal, obtaining a bandwidth expansion high frequency component, and overlapping the low frequency signal with the bandwidth expansion high frequency component for outputting. According to the method and the device, the bandwidth expansion processing mode is adjusted adaptively, the bandwidth expansion processing is carried out on the narrowband voice signal, the voice quality is improved, the broadband voice signal is outputted directly, the wrong expansion of the broadband voice signal is avoided, and the voice quality is ensured further.

The invention relates to a dynamic register method of IP sub-telephone roaming, wherein it arranges sub-telephone attach server address on the terminal device of sub-telephone; the terminal device registers into said server; the server provides available server address list to the terminal device; when the sub-telephone roams, the terminal device will automatically process IP connection test and voice quality test with all available server at background, and the test result will be evaluated via the response delay and loss rate of network connection, while the top one has best network condition; then the terminal device selects the best server to dynamically register; the server processes register refresh of soft exchange and service route. The invention can automatically register without changing the number of sub-telephone, to improve service quality, used in IP-FBX server and video communication system.

The invention relates to a method for initiating encryption during call establishing procedure and a wireless network controller. The method includes receiving a message of that a call has been established; determining whether the present time is longer than the encryption activating time provided in the message; and if yes, initiating encryption using the exceed frame number in the message and the present connecting frame number at network side as the encryption parameters. Synchronization of encryption parameters of RNC and the encryption terminal is ensured, therefore flow noise of audio is prevented and the audio quality is further ensured. The wireless network controller includes a message receiving module, a time determining module and an encryption initiating module to ensure synchronization of encryption parameters of RNC and the encryption terminal, therefore flow noise of audio is prevented and the audio quality is further ensured.

The invention relates to a GAN-based speech confrontation sample generation method, which is characterized by comprising the steps of preprocessing an original speech data sample x, inputting the preprocessed original speech data sample x into a generator G to obtain an adversarial disturbance G(x), and using a formula (1) to construct an adversarial sample, the formula (1) being xadv = x + G(x),inputting the adversarial sample xadv into a discriminator D, and inputting the adversarial sample xadv into a target network f after the adversarial sample xadv passes through a Mel-frequency cepstrum coefficient MFCC feature extractor, calculating the loss lf of the target network, the adversarial loss lGAN of the discriminator, the hinge loss lhinge, the mean square error loss l2 and the loss lD of the discriminator, thereby obtaining a loss function l when the generator G is trained, S4, updating parameters of a generator and a discriminator through gradient back propagation of the loss function l obtained in the S4, obtaining an optimal generator through a formula (10), loading an original sample x into the optimal generator obtained in the S5 through the formula (10), and constructing to obtain a corresponding adversarial sample. Thus, the minimum disturbance can be effectively generated, and the speech quality can be ensured.

The invention discloses a method for realizing communication denoising. The method is applied to a terminal, and the terminal is equipped with at least one auxiliary terminal. The method comprises the steps that the terminal, which is in a communication state, obtains first communication signals in the communication through a microphone of the terminal; second communication signals acquired by a microphone of each of the at least one auxiliary terminal in the communication are obtained; an auxiliary microphone of the terminal is determined according to the second communication signals acquired by the microphone of each auxiliary terminal; and voice signals of the terminal in the communication are separated from the first communication signals and the second communication signals acquired by the auxiliary microphone. Meanwhile, the invention further discloses the terminal.

The invention provides an audio mixing method, a server and clients. The audio mixing method comprises the steps that first audio signals sent by all the clients are received; the received first audiosignals are subjected to full audio mixing processing to obtain full audio mixing audio signals; and the full audio mixing audio signals are sent to all the clients, so that the clients filter the first audio signals subjected to full audio mixing processing from the full audio mixing audio signals so as to obtain audio mixing audio signals carrying the first audio signals, subjected to full audio mixing processing, of other clients to be played. Through the audio mixing method, the first audio signals of the clients on all channels are subjected to audio mixing only through one-time audio mixing processing, and the cost of audio mixing processing is lowered greatly while the audio quality is ensured.

The present invention discloses a control method of the quality of a service supporting inter-office IP and a TDM mixed network. The method is used for a communication system consisting of a switching controller and a media gateway. The method comprises a service quality information monitoring procedure, a service quality information reporting procedure and a load sharing procedure, wherein the service quality information monitoring procedure is used for the media gateway to monitor the called service quality index during the current network running process; the service quality information reporting procedure is used for the media gateway to report the called service quality index monitored to the switching controller; the switching controller makes statistics for the service quality index, and adjusts the load sharing proportion according to the statistical service quality condition. The present invention also discloses a communication system for applying the control method of the quality of the service supporting the inter-office IP and the TDM mixed network. The application of the present invention can effectively realize the reasonable use of the IP network and the TDM resources, and can ensure the speech quality.

The invention relates to the field of intelligent communication equipment, in particular to an intelligent sound box system with a telephone function, which comprises an intelligent sound box, an intelligent mobile terminal and a cloud server, the intelligent sound box comprises a Wi-Fi module, a microphone array, an MCU, an AEC module, an AGC module and an ANS module. The intelligent mobile terminal comprises an address book management module and a network distribution unit; the cloud server comprises a voice server, an address book management unit and a telephone number management unit; theWi-Fi module is connected with the network distribution unit for network distribution and can access the cloud server after accessing the home network router, and the intelligent mobile terminal address book management module uploads address book information to the cloud server address book management unit; the microphone array is used for pickup and monitoring a user instruction; and after receiving the instruction, the MCU can call the AEC module, the AGC module and the ANS module to process the audio data. The terminal equipment has the beneficial effects that a fixed-line telephone is replaced, the calling function is realized, the interaction function is also realized, and the convenient calling terminal equipment is provided.

The invention discloses a voice control method of a networking telephone sending terminal, comprising the following processes of collecting, coding and sending voice data, wherein the coding process comprises the step of dynamically regulating a coding period according to occupation conditions of a coding buffer area and a sending buffer area. The invention also comprises a networking telephone sending terminal. The voice control method dynamically regulates the coding period according to the occupation condition of the coding buffer area, avoids the problem that terminal resources are limitedto cause the result that voice data can not be coded in time and can be covered, thereby the voice quality of the networking telephone communication is ensured.

The invention discloses a business hierarchical processing method and device for VoLTE voice business and data business, electronic equipment and a computer storage medium, and the method comprises the steps: receiving a business request transmitted by user equipment; determining a business type corresponding to the business request according to the business request; if the business type is a bigpacket data business, switching the user equipment to a first frequency band, and bearing the service corresponding to the service request of the user equipment by using the first frequency band; andif the service type is a VoLTE voice service or a packet data service, switching the user equipment to a second frequency band, and bearing the service corresponding to the service request of the userequipment by using the second frequency band. According to the scheme, the VoLTE voice service and the data service are subjected to hierarchical processing, the first frequency band is used for bearing the large packet data business, and the second frequency band is used for bearing the VoLTE voice service and the small packet data service which have relatively small requirements on the RB, so that the voice quality of the VoLTE voice service is effectively guaranteed, and voice access is smoother.

The embodiment of the invention discloses a data packet decompression method and device. When the packet header of the current data packet fails to be decompressed; whether packet header decompressionfailure occurs in a conversion process between a call period and a silent period or not is determined; and when packet header decompression fails and occurs in a conversion process between a call period and a silent period, obtaining a plurality of timestamps to be verified according to the message serial number difference value, the timestamp step length of the data packet decompression contextand the timestamp of the latest correctly decompressed data packet, and updating the context by using the correct timestamp of the CRC in the plurality of timestamps to be verified. According to the embodiment of the invention, decompression failure caused by key data packet loss or compatibility problem of a compression coding mode can be reduced, so that continuous packet loss or repeated packetloss caused by decompression failure is avoided, and the voice quality is ensured.

The invention provides a silent upgrading method and equipment for earphones and a charging box. The silent upgrading method comprises the following steps: establishing wireless communication connection between a main earphone and a mobile terminal; receiving firmware upgrading files sent by the mobile terminal at different transmission rates according to different working states of the main earphone; enabling the slave earphone to receive a firmware upgrading file the same as that of the master earphone; enabling the master earphone and the slave earphone to perform firmware upgrading according to the firmware upgrading files, and enabling the charging box to inquiry whether the firmware upgrading files in the data storage areas of the two earphones are updated or not; and if so, obtaining the updated firmware upgrading file from the earphone, and carrying out firmware upgrading on the charging box by utilizing the updated firmware upgrading file. According to the silent upgrading method and equipment for earphones and charging box, the time of the user is not occupied in the upgrading process, silent upgrading which is not perceived by the user is achieved, the transmission ratecan be automatically adjusted according to the current service state of the earphone when the firmware upgrading file is transmitted, the audio quality is not affected, and the user experience is improved.

The invention provides a single-card multi-module terminal and a communication processing method thereof, and belongs to the field of mobile communication. The single-card multi-module terminal comprises a master processing unit and a slave processing unit, wherein the master processing unit comprises an SIM (Subscriber Identify Module) driving module, an SIM access agent module and an SIM access module; the SIM access agent module is connected with the SIM driving module and the SIM access module, and is used for receiving access request information of the SIM access module from the master processing unit or the slave processing unit, sending the access request information to an SIM card through the SIM driving module and sending access response information to the corresponding SIM access module after the access response information from the SIM card is received. According to the technical scheme of the invention, the service delay of the single-card multi-module terminal can be reduced, incoming calls are prevented from being lost, and the utilization rate of 3G (third Generation) network is improved.

The invention discloses an echo cancellation method and device, which are used for solving the technical problem of poor echo cancellation effect. The method comprises the steps that an echo cancellation device in the video conference system determines the distortion level of audio equipment in the video conference system, the audio equipment comprises playing equipment and collecting equipment, and the distortion level is used for representing the nonlinear distortion degree of audio data played by the playing equipment and collected by the collecting equipment; a target suppression level corresponding to the distortion level is determined, the target suppression level is used for representing the degree of nonlinear echo suppression processing on the audio data; and non-linear echo suppression processing is performed on the audio data acquired by the acquisition equipment through the target suppression level to obtain the audio data after the non-linear echo suppression processing.

The invention belongs to the technical field of communication, and discloses a VoLTE accurate switching method and system based on user perception, and the method comprises the steps: determining a KPI index and a convergence threshold of a perception dimension through the hierarchical mapping of a KQI and a KPI index according to the perception dimension of a user, building a KPI difference portrait model, and carrying out the comprehensive scoring of a small service and a target cell; if the total weight score of the serving cell exceeds the total weight score of the neighbor cell, startingan emergency switching process; and when the serving cell does not reach the condition of starting the emergency switching threshold, namely the total weight score is smaller than the total weight score of the adjacent cell, performing pilot frequency inter-system switching based on A2 + A4 / A2 + A5 / A2 + B2 according to a normal switching process. According to the invention, the emergency switchingof the perception dimension is introduced to enable the UE terminal to quickly start the emergency switching when the perception of the serving cell is reduced, thereby effectively avoiding the problems of packet loss, interruption, single pass, call drop and poor MOS quality caused by the degradation of the UE terminal in the aspects of interference, quality, load, distance and the like in the prior art, and improving the voice quality.

The invention discloses a coding method and device of a digital mobile radio (DMR) system, a storage medium and a digital radio. The coding method comprises the steps: a voice signal is sampled, quantified and coded to form subframes, the subframes comprise a plurality of characteristic parameters, the multiple characteristic parameters comprise the pitch period, line spectrum frequency coefficients, energy and unvoiced or voiced sound discrimination, and the one or more characteristic parameters are obtained through code book quantification; the subframes with the preset number are spliced toform a voice frame; and the voice frame is subjected to forward error correction to obtain a coding frame. Through the technical scheme, coding bits of the characteristic parameters can be compressed, redundant bits of forward error correction can be increased, the noise resistance of coded data can be enhanced, and the voice transmission quality is improved.

The invention discloses a UMTS mobile terminal circuit domain voice encryption communication technology realization method. The UMTS mobile terminal circuit domain voice encryption communication technology realization method comprises the following steps of S1, a system hardware configuration design; S2, a system software configuration design; S3, protocol signaling level realization; S4, network mode control realization; S5, voice encryption communication flow realization; S6, terminal encrypted conversation calling flow; and S7, a terminal called flow. In the design, a control level and a data level are borne in a voice channel band; secret key negotiation and voice frame encryption and decryption logic is completed in a TPM hardware unit and a TEE execution environment so that safety of a secret key, an algorithm and a system execution environment is ensured and safety of voice communication during source storage, processing and transmission is ensured too; and a voice flow is ensured not to carry out any voice coding and decoding conversion during air interface relay transmission and voice tone quality is guaranteed.

The invention relates to a speech feature enhancement post-filtering method based on a deep neural network, and belongs to the technical field of speech filtering, and the method comprises the following steps: S1, mixing pure speech with noise according to different signal-to-noise ratios, and generating training data; S2, selecting the logarithm power spectrum LPS of the training data as a feature for extraction, and taking the LPS of the pure voice as a target; S3, performing training by using a standard structure deep neural network DNN; S4, performing loss estimation on a training result;and S5, inputting a loss voice, and performing loss compensation based on loss estimation. Compared with the prior art, noise interference can be effectively suppressed while the voice quality is ensured.

The embodiment of the invention provides a data processing method and device, and the method comprises the steps of a service cell receiving a voice call establishment request sent by a user terminal,and determining a frequency point corresponding to the service cell; judging whether the frequency point corresponding to the serving cell is a specified frequency point or not, the interference degree of the specified frequency point being smaller than that of a non-specified frequency point; if the frequency point corresponding to the serving cell is not the specified frequency point, generating a corresponding switching instruction according to the specified frequency point and sending the switching instruction, the switching instruction being used for indicating that the user terminal accesses the cell corresponding to the specified frequency point; And if the frequency point corresponding to the serving cell is an appointed frequency point, generating a corresponding A5 event controlinstruction according to the non-appointed frequency point and sending the A5 event control instruction, so that the user terminal resides in the serving cell. Therefore, the voice quality is ensuredby ensuring that the cell accessed by the user terminal performing the voice service is the cell corresponding to the appointed frequency point.

This invention provides a method for realizing hairpin link including: a first context adds the terminal point in a second context with the hairpin link relation into itself, then sets up a hairpin link relation in the first one, the second one copies the TDM end point, grouping end point and end point connection information of the second user to the first one, which cuts off the connection between its own first user TDM point and its grouping end point then cuts off the copied connection between the second user TDM end point and its grouping point, then the first one connects its TDM end point with the copied second user TDM end point to realize hairpin link. In this invention, hairpin link can be existed in the first and the second contexts, yet, if the MGC does not make a correct estimation, the MG can realize hairpin link itself.

The present invention pertaining the mobile communication field discloses a channel transmit limiting method and the user terminal to ensure the voice quality of the half speed service channel. In the present invention, the transmitting of GPRS signaling data of PS domain on the main special control channel is forbidden when the user terminal transmits data through the half speed service channel. The LAPDm biggest frame sends by the network side is only used when the user terminal transmits data through the full speed service channel and the transmitting of the LAPDm frame on the main special channel in the special mode is limited.

The invention relates to a packet transmit-receive system, comprising a detection module, an abandoning rate determining module, a transmitting module and a reckoning module; wherein, the detection module is used for detecting the network choking condition. The abandoning rate module determining module is used for determining the packet abandoning rate according to network choking condition. According to the packet abandoning rate, the transmitting module is used for evenly abandoning one part of a plurality of voice packets which are to be transmitted and transmitting the voice packets which are not abandoned. The reckoning module is used for receiving the voice packets which are not abandoned and reckoning the voice packets which are abandoned. The invention also provides the packet transmit-receive method. The packet transmit-receive system and the method determine the packet abandoning rate according to the network choking condition, evenly abandon the voice packets according to the packet abandoning rate, hence can effectively reduce current the capacity of the network and can guarantee the voice quality of the network voice.

A head-mounted intelligent voice recognition device disclosed by the present invention comprises a fixing device, a supporting device, a vibration damping device and a sound pick-up, the fixing device is movably connected with the supporting device, one side of the supporting device is fixedly connected with one side of the sound pick-up, the vibration damping device is connected with the fixing device, and a vibration sensor is arranged on the vibration damping device. The vibration sensor is connected with the vibration damping device, the fixing device comprises a fixing frame, fixing seats are arranged at the two ends of the fixing frame respectively, one side of each fixing seat is fixedly connected with one side of the fixing frame, and a first rotation groove and a second rotation groove are formed in each fixing seat in sequence, the supporting device comprises a first rotation frame and a second rotation frame, and first ball heads are arranged at the two ends of the first rotation frame respectively. The first reversing frame is fixedly connected with the first ball head, the outer circle face of the first ball head is movably connected with the groove face of the first reversing groove, the second ball heads are arranged at the two ends of the second reversing frame respectively, and the second reversing frame is fixedly connected with the second ball heads.

The invention discloses an audio noise reduction filtering method, a noise reduction filter device, electronic equipment and a computer storage medium, and relates to the technical field of audio signal processing. The method comprises the following steps: acquiring characteristic parameters of an audio input signal by using a preset neural network; calculating a filtering weight coefficient based on the characteristic parameters; processing the audio input signal based on the filtering weight coefficient to obtain a filtered audio signal; calculating a cost value based on the filtered audio signal and the real signal; and training a preset neural network by using the cost value. Through the above mode, the audio noise reduction filtering method can effectively reduce the noise in an audio system and improve the voice quality.