30results about How to "Good voice effect" patented technology

In frequency signals obtained by converting input audio signals from time-domain signals to frequency-domain signals, a level control value setting unit 5 establishes a level control value for reducing the levels of spectrums at a noise-components level. A level control value smoothing unit 6 carries out a smoothing process of smoothing the level control value established by the level control value setting unit 5 temporally. A spectral adjustment unit 8 multiplies the level control value after the smoothing process by the frequency signals, performing a level control.

The embodiments of the application provide an echo cancellation method and a terminal, which relate to the technical field of real-time audio and video communication. The method includes the followingsteps: a terminal collects first-end audio data, wherein the first-end audio data includes the voice of a first-end user and the audio played by an audio playback device on the terminal; reference audio data corresponding to the first-end audio data is queried from a cache region, and the cache region caches audio data to be played on the audio playback device as reference audio data; the audio played by the audio playback device in the first-end audio data is eliminated by using the reference audio data, and corrected audio data is determined; and the corrected audio data is sent to a second-end user terminal. Because the audio played by the audio playback device in the first-end audio data is eliminated by using the reference audio data and the voice of the first-end user is retained, the interference of the audio played by the audio playback device to the voice of the first-end user is avoided, and the quality of communication between the first-end user and the second-end user is improved.

This invention discloses a mixture sound process method, which comprises the following steps: orderly extracting each sample point vibration value from at least two paths of impulse coding modulation PCW flow needed for mixture process; b, using preset weight value to process weight on sample point vibration range for each path of PCM flow and adding vibration range to get value as mixture PCW flow sample vibration range; c, orderly coding each sample point of PCM flow to change into mixture PCM flow.

The invention discloses a method and a device for handling receiving voice, which belongs to the technical field of communication. The method comprises the steps of: acquiring the use condition information of a voice output device; respectively regulating the signal intensity of the receiving voice received from each frequency band according to the use condition information of the voice output device, and outputting the regulated receiving voice through the voice output device, wherein the frequency bands are obtained by dividing the voice bandwidth of the voice output device. The device comprises an acquisition module and a handling module. In the invention, by acquiring and in accordance with the use condition information of the voice output device, the signal intensities of the receiving voice received from each frequency band are regulated respectively, the signal intensity of voice in a frequency band with higher signal intensity attenuation is intensified greatly, the signal intensity of voice in a frequency band with lower signal intensity attenuation is intensified slightly, and thus, the problems that the receiving voice is sharp and the volume is low caused by high signal intensity attenuation of the voice in a low frequency band in the process of using a mobile phone can be solved, and the integral voice effect of the receiving voice can be improved.

Embodiments of the present invention relate to the technical field of image processing, and in particular, to a video conference management method and device. The method is applied to an electronic device communicatively connected with multiple cameras, each camera is set on a conference table in a meeting place, the conference table is provided with a microphone, and the microphone is communicatively connected to the electronic device. The real-time images are spliced ​​to obtain a real-time panoramic image of the venue. According to the real-time panoramic image, it is judged whether at least one participant among the multiple participants has the behavior of triggering the microphone for voice collection. If at least one participant has the behavior of triggering the microphone for voice collection Behavior, control the microphone to turn on, receive the real-time voice information collected by the microphone, and send the real-time panoramic image and real-time voice information synchronously. The method can improve the intelligent degree, voice effect and interactive feeling of the video conferencing system.

A dual-microphone arrangement (300) provides improve voice performance in a wireless headset (12). A vibration sensor (1130) is used for voice pickup and will add low-frequency voice audio content in windy conditions. An equalizer (810) is used to restore low-frequency voice audio content in wind-free conditions. Depending on the measured wind power, the output will derive more signal from the equalizer (810) or more signal from the vibration sensor (1130).

The present application discloses a method, a device and a server for implementing a stereo call on an ESCO link, belonging to the technical field of voice processing. The method for implementing a stereo call on an ESCO link includes: establishing Bluetooth communication between a microphone and an intelligent terminal device through an ESCO link; receiving vocal data and accompaniment data from the intelligent terminal device through the microphone; The data and the accompaniment data are mixed and processed, and the result of the mixing processing is sent to another intelligent terminal device. The application of the present application improves the effect and quality of calls and improves user experience.

The application discloses a data processing method, device, electronic equipment and storage medium, and relates to the technical fields of voice interaction technology, automatic driving, smart home and deep learning. The specific implementation scheme is: obtain 16-bit model data to be processed; obtain a 4-bit load instruction; load four 4-bit model data from the 16-bit model data into a 64-bit register according to the 4-bit load instruction . Therefore, the present application can load four 4-bit model data from 16-bit model data into 64-bit registers by adding a 4-bit load instruction, thereby improving the processing efficiency of low-bit model data and effectively reducing network communication. The sequential access throughput caused by the cost cannot be improved and broken through. Furthermore, by adding custom instructions, the chip can store a larger neural network without increasing the chip area to achieve better voice effects.

The present invention discloses an analog voice handle for testing. The analog voice handle comprises a voice coding and decoding chip, a permalloy audio transformer, a T-type attenuator and an optocoupler; two sets of transformers are arranged in the permalloy audio transformer; the impedances of the two sets of transformers are 150: 150 and 600: 600 respectively; one terminal of the input end of each of the two sets of transformers is connected with system ground; one terminal of the output end of each of the two sets of transformers is connected with voice handle ground; the non-grounding ends of the two sets of transformers are an MIC lead and a PHONE lead respectively; the voice handle ground and the system ground are connected with each other through a capacitor; the T-type attenuator includes resistors R1, R2 and R3; the first end of the resistor R2 is connected with the AO end of the voice coding and decoding chip; the second end of the resistor R2 is connected with the first end of the resistor R1 and the first end of the resistor R3; the second end of the resistor R3 is grounded; the second end of the resistor R1 is connected with the non-grounding end of the input end of the 150: 150 transformer in the permalloy audio transformer; and the non-grounding end of the input end of the 600: 600 transformer in the permalloy audio transformer is connected with the AI end of the voice coding and decoding chip; and one output end of the optocoupler is a P TT lead.

The embodiment of the present invention provides a microphone matrix adjustment method, and the method comprises the steps: determining the shape of a socket matrix board according to a preset position of a target product; setting at least one microphone unit on a socket matrix board according to a voice algorithm to obtain a microphone matrix; detecting the microphone matrix, and adjusting the position and the number of the microphone units until a preset condition is satisfied when it is detected that the microphone unit does not meet the preset condition. The microphone device and the audiodevice are also disclosed in the embodiments of the present invention. Since the microphone matrix in the embodiment of the present invention is divided into two parts of the socket matrix board andthe microphone units, and the microphone units can be flexibly disposed on the socket matrix board, the position, number and relative angles of microphone units can be easily adjusted when detecting the microphone matrix, so as to flexibly cooperate with the voice algorithm, thereby meeting different product requirements, greatly reducing the cost of research and development products, and improving design efficiency.