85 results about "Activation detection" patented technology

The present invention concerns phrenic nerve activation detection algorithms for characterization of phrenic nerve activation and phrenic nerve activation avoidance in cardiac pacing therapy.

Various embodiments concern receiving a respiration signal indicative of respiratory activity of the patient, identifying respiratory phases based on the respiration signal, delivering cardiac pacing pulses within each of the identified respiratory phases, receiving a phrenic nerve activation signal indicative of activation of the patient's phrenic nerve, analyzing the phrenic nerve stimulation signal to determine if one or more of the pacing pulses activated the phrenic nerve of the patient, and determining if at least one of the delivered pacing pulses activated the phrenic nerve of the patient based on the phrenic nerve activation signal indicating activation of the patient's phrenic nerve associated with delivery of the at least one cardiac pacing pulse.

An implantable cardiac device includes a sensor for sensing patient respiration and detecting phrenic nerve activation. A first filter channel attenuates first frequencies of the sensor signal to produce a first filtered output. A second filter channel attenuates second frequencies of the respiration signal to produce a second filtered output. Patient activity is evaluated using the first filtered output and phrenic nerve activation caused by cardiac pacing is detected using the second filtered output.

The invention relates to a pick-up method and system based on a microphone array; the method comprises the steps of 1, using the microphone array to pick up and output one of multiple speech signals for speech activation detection, and judging whether a speech activation signal occurs or not; if yes, executing step 2; if not, repeating the step 1; 2, performing sound source positioning on the multiple speech signals to obtain a sound source positioning direction; 3, subjecting the speech signal of the sound source positioning direction to speech augmentation to obtain an augmented speech signal; 4, subjecting the augmented speech signal to speech wakeup detection, and judging whether speech wakeup is detected or not; if yes, executing step 5; if not, repeating the step 1; 5, allowing the microphone array to pick up and output the multiple speech signals; 6, processing the multiple speech signals as one signal that is output finally as picked speech. Speech signals can be better picked up in far field environment, particularly accurately in high-noise environment, and an excellent solution is provided for remote speech control.

The embodiment of the invention provides a voice wake-up method and apparatus. The method comprises: periodic sampling is carried out on an audio signal, wherein sampling is carried out at ti time to obtain a sampling signal; audio energy of the sampling signal is calculated; when the audio energy is larger than or equal to a first threshold value at the ti time, a digital signal processor (DSP) is waken up to carry out voice activation detection (VAD); when the VAD fails, detection fails n times continuously before the ti time, and a difference value between first noise energy and a first threshold value at the ti time is larger than a preset first threshold value, a second threshold value is generated according to the first noise energy and is used as a first threshold value at ti+1 time, wherein the first noise energy is obtained by extracting the sampling signal by a first extraction rate 1/x and carrying out slow-speed tracking filtering on the extracted sampling point. According to the embodiment of the invention, the number of times of VAD is reduced and reduction of power consumption of the terminal in a noisy environment is realized.

The present invention provides a method for detecting active code channels of multiple users in a wireless communications system. The method comprises the steps of: receiving the radio signals that contain the signals of at least two of said multiple users; separating the data and training sequences of said received radio signals; making channel estimation based on said separated training sequences to obtain the estimated channel response; obtaining, based on said estimated channel response, the active channel response windows; obtaining the allocated spreading codes to which said active channel response windows may correspond; determining, based on the estimated channel responses and the possible corresponding allocated spreading codes, the active spreading code channels corresponding to said active channel response windows; making multi-user detection in the determined active code channels. Thus the blind or semi-blind detection of the receiver is implemented.

The invention discloses a multi-activation detection (MAD) method for an intelligent elastic framework (IRF), which comprises the steps of: receiving MAD messages at least containing an up-link bandwidth and being sent by other main devices by a main device; closing a service port of the device and service ports of the other devices of the IRF belonging to the device when any up-link bandwidth accepted by the main device is larger than the up-link bandwidth of the IRF belonging to the device; and maintaining the normal operation of the IRF belonging to the device when the up-link bandwidth accepted by the main device is smaller than the up-link bandwidth of the IRF belonging to the device. On the basis of the same invention concepts of the method, the invention also provides a device capable avoiding the breakdown status of all the services or most services of a stackable system.

The invention provides a mute indication method and device applied for a video conference. The method comprises: a multipoint conference unit (MCU) carries out sound activation detection for a voice frequency media flow sent by a terminal which participates in the video conference and is muted; the MCU obtains the detection result of the terminal, wherein the detection result comprises one of the following states: sound activation state and sound inactivation state; and when the detection result is the sound activation state, the MCU superposes the mute video indication to the video signal of the terminal. The invention can improve the communication experience of the video conference and makes the video conference simple and efficient.

The invention relates to voice activation detection (VAD), and a method and apparatus for the VAD. The method comprises: obtaining the sub-band signal and the frequency spectrum amplitude of a current frame; according to the sub-band signal, calculating to obtain the frame energy parameter of the current frame and the value of a spectrum gravity center characteristic parameter; according to the background noise energy obtained through estimation of a previous frame, the frame energy parameter of the current frame, and signal-to-noise ratio sub-band energy, calculating to obtain the signal-to-noise ratio parameter of the current frame; and according to a tonal sign, the signal-to-noise ratio parameter, the spectrum gravity center characteristic parameter and the frame energy parameter, calculating to obtain a VAD determination result. The method and apparatus provided by the invention can improve the detection accuracy of unstable noise (such as office noise) and music.

Disclosed in the present invention are a transmission method and apparatus based on channel activation detection, relating to the field of communications. The method comprises: a base station issuing to a terminal scheduling information containing scheduling resource indication, for enabling the terminal to send service data; the base station receiving the service data uploaded by the terminal; the base station verifying the service data; in the event of a verification error, the base station judging whether the scheduling information is lost by using the service data; and the base station performing HAQR (Hybrid Automatic Repeat request) mode decision processing and HARQ retransmission processing by using a judgment result, for controlling the terminal to perform HARQ retransmission according to a processing result. By adopting the transmission method and apparatus disclosed by the invention, Long Term Evolution (LTE) uplink packet error can be reduced, and the resource utilization rate of a physical downlink control channel (PDCCH) can be effectively improved.

The invention relates to a field of electronic technology and specifically relates to an awakening device and method. The voice activation detection based awakening device includes a microphone used for sensing sound signals and converting the sound signals to electric signals; a first signal extracting unit connected to the output terminal of the microphone and used for extracting base frequency signals in the output signals of the microphone; a comparison unit connected to the first signal extracting unit and comparing the base frequency signals with a reference base frequency range indicating voice features and generating a first comparison result; and a controller connected to a comparison unit and producing an awakening signal under the effect of the output signals of the comparison unit. According to the invention, the device can identify the distance of a sounder besides recognizing whether the voice signals are consistent to features of human voice or not, so that interference on normal voice activation detection due to irrelevant voice signals can be avoided and voice awakening can be realized effectively.

The invention discloses a signal processing method and a signal processing device for a microphone array. The method comprises the following steps: fixed beam-forming is carried out on M input signals acquired by a microphone array to obtain reference target signals, voice activation detection is carried out on the reference target signals to obtain a voice activation detection result, target appearing probability analysis is performed on the M input signals to obtain the target appearing probability, and the M input signals are passed through a congestion matrix to obtain (M-1) reference interference signals; the filter coefficient of a filter bank for carrying out filtering is determined according to the voice activation detection result and the target appearing probability, and the (M-1) reference interference signals are filtered by using the determined filter coefficient of the filter bank to obtain filtered reference interference signals; and the filtered reference interference signals are subtracted from the reference target signals to obtain output signals of the microphone array. Damage to a target signal can be minimized.

The invention relates to the test technology for network communication devices and discloses a system for testing the multi-activation detection performance of a stack system accurately and efficiently. The system comprises a tester, the stack system composed of two tested devices and two auxiliary test devices, wherein the two tested devices of the stack system are connected through a virtual exchange link and in physical connection with the two auxiliary test devices respectively; and the two auxiliary test devices are connected with a test port of the tester respectively. In addition, the invention further discloses a method for testing the multi-activation detection performance of the network communication devices correspondingly.

The invention discloses a method and equipment for controlling information activation detection. The method comprises the following steps: determining a detection signal and a modulation constellation point of the detection symbol and control information; and according to the detection signal and the modulation constellation point of the detection symbol and the control information, judging whether control information is borne or not. According to the method and the equipment, the virtual detection probability of the control information can be effectively lowered.

The invention provides a new self-adapting multi-speed coder and a coding method. Compared with prior art, the voice activation detection is largely changed in that the voice activation detection is aimed at the synthesized digital sound corresponding to the coding frame, and the overall framework of the self-adapting multi-speed coder and the coding method is renewed, so the sound signal synthesized by the coder can accurately reflect the audio effect of the original sound. The invention can be directly applied to voice encryption technique of the third generation of mobile communication system-Universal Mobile Telecommunication System.

The invention features the following: using a calling model to simulate the call interaction between both calling sides; calculating the state of call process at each time point; according to the sound activation detection technology, deciding if said state generates the traffic of network voice call; at the time point of generating the traffic, recording the offset between the time point and the initial time, the message sequence number and the size of the message corresponding to the code to form a traffic sequence; using the interaction between two test ends to determine the start time and the message actually-sending time, and finally generating the network voice call test traffic based on calling model.

The invention discloses a pickup method based on a microphone array, and the method comprises the steps: using one of a plurality of voice signals, picked up and outputted by the microphone array, forvoice activation detection; performing the sound source positioning through the multi-channel voice signals outputted by the microphone array to obtain a sound source positioning direction; performing voice enhancement of the voice signal in the sound source positioning direction to obtain an enhanced voice signal; performing voice wake-up detection of the enhanced voice signal; processing the multi-channel speech signals picked up by the microphone array into an enhanced speech signal as the final picked-up voice output. The present invention also discloses a pickup system based on the microphone array, an electronic device, and a computer readable storage medium. The invention can better pick up the voice signals in a far-field environment through the voice activation detection and thevoice wake-up through the speech enhancement, especially in a high noise environment.

An approach is provided that receives an audio stream and utilizes a voice activation detection (VAD) process to create a digital audio stream of voices from at least two different speakers. An automatic speech recognition (ASR) process is applied to the digital stream with the ASR process resulting in the spoken words to which a speaker turn detection (STD) process is applied to identify a number of speaker segments with each speaker segment ending at a word boundary. A speaker clustering algorithm is then applied to the speaker segments to associate one of the speakers with each of the speaker segments.

The present invention discloses an activation code channel detecting method which is applied on the multiple-antenna combined-detecting system, and the method comprises the following steps: A. executing channel estimating window activating detection according to the multiple-antenna channel estimation after the noise threshold detection; B. executing matched filtering respectively to each antenna according to the activation detecting result of the channel estimating window, summating the symbol of same code channel of multiple-antenna at a same window position according to the result of matched filtering, and executing interference-user non-preferred code channel activation detecting according to the summated symbols; and C. executing combined detection interference eliminating to the summated symbols according to the activation detecting result of the interference-user non-preferred code channel, and executing non-preferred code channel activation detection to the symbol executed with interference elimination. The method according to the invention leads to the reducing of the demodulating threshold at the state of no increasing the band width thereby increasing the system throughout, and at the same time the coverage of the service can be increased.

The invention provides a voice quality assessment method and a device based on voice activation detection. Framing processing is performed on an input sample audio file and a short-time stationary audio signal is acquired. The voice activation detection is performed on the short-time stationary audio signal and a frame corresponding to the short-time stationary audio signal is determined to be a voice frame or a non-voice frame. A preset frame number is taken as a step length so as to carry out voice quality detection on fragment audio files of a sample audio file, the fragment audio files with qualified quality are reserved and the unqualified fragment audio files are abandoned. When only a voice activation detection mode is combined, voices and noises are distinguished, but voice qualitycan not be assessed so that input voice quality can not be quantitatively controlled. In the invention, the above technical problem is solved.

A data transfer memory for reducing the number of components in an electronic module. A master controller circuit provides a transfer start command to a master clock signal generator circuit when receiving an activation detection signal from a power activation detection circuit. As a result, the master clock signal generator circuit generates a basic clock signal, outputs the basic clock signal to an SCL line, and has a master transfer sequencer circuit execute a transfer sequence. The master transfer sequencer circuit transmits a start condition, data stored in the nonvolatile memory via a serial control circuit, and a stop condition to an SDA line synchronously with the basic clock signal.

In a system and computer implemented method for mapping of an anatomic structure and bi-directional activation detection of electrograms such as atrial and/or ventricular electrograms, both positive and negative deflections of an electrogram signal are analyzed over an analysis time period of the signal. At least one characteristic of the electrogram signal is determined based at least in part on analyzing both positive and negative deflections of the signal over the analysis time period. The determined at least one characteristic of the atrial electrogram signal is then associated with a generated three-dimensional model of the anatomic structure.