Huawei Technologies Co., Limited

Huawei Device Co., Ltd.

Dongqi Wang

Wuzhou Zhan

Lei Miao

Qing Feng Zhang

Fengyan Qi

Yi Yang

Jianfeng Xu

Chen Hu

Zhengzhong Du

Jing Wen Wang

Wei Li

Huangwei Wu

Lei Li

Zhiming Ding

Qing Zhang

Jing Wang

Yongfeng Tu

Lijing Xu

Pulin Wang

Zhihui Liu

Xiaoxia Wei

Yanbo Long

Ningbo Qiao

Weishan Chen

Junjun Liao

Cheng Hu

A method and an apparatus for calling a terminal to join a conference, related to the field of communications technologies, resolving a problem that a process in which a calling terminal calls a called terminal to join the conference or a called terminal requests joining the conference is complicated. The method includes the following steps: receiving first call request information, where the first call request information carries identification information of the called terminal; and calling the called terminal according to the identification information of the called terminal, so that the called terminal receives a message which is published by another terminal in a current conference and that the called terminal publishes a message to another terminal in the current conference. The method may be applied to calling a terminal to join a conference.

Method and apparatus for calling terminal to join conference

An audio mixing method, apparatus and system, which can ensure sound quality after audio mixing and reduce consumption of computing resources. The method includes: receiving an audio stream of each site, and analyzing the audio stream of each site to obtain a sound characteristic value of a sound source object; selecting, according to a descending sequence of sound characteristic values of sound source objects, a predetermined number of sound source objects from the sound source objects to serve as main sound source objects; determining, according to a relationship between a target site and the sites where the main sound source objects are located, audio streams that require audio mixing for the target site; and performing audio mixing on the audio streams that require audio mixing for the target site and sending the audio streams after the audio mixing to the target site.

Audio mixing method, apparatus and system

A method and an apparatus for playing audio of an attendant at a remote end and a remote video conference system are provided. The method includes: receiving audio of an attendant at a remote site; and by means of two or more loudspeakers mounted at a top and a bottom of a remote image presentation device at a local site, simulating an audio transmission path of the audio of the attendant at the remote site between the two or more loudspeakers and an attendant at the local site through a predetermined algorithm, where the simulating means that the audio transmission path is simulated between an image of the head of the attendant at the remote site displayed in the remote image presentation device, and the head of the attendant at the local site.

Method and apparatus for playing audio of attendant at remote end and remote video conference system

The present invention discloses a signal processing method, system and apparatus for 3-dimensional (3D) audio conferencing. The implementation is: a server obtains at least one audio stream relative to one terminal; the server allocates identifiers for the obtained at least one audio stream relative to the terminal; and the server combines the obtained at least one audio stream and the identifiers of the at least one audio stream and sends the combination to the terminal. With the technical solution of the present invention, the issue of excessive transmission channels required in the prior art is resolved and the terminal is capable of determining the sound image positions of other terminals freely.

Signal processing method, system, and apparatus for 3-dimensional audio conferencing

A conference terminal, a conference server, a conference system, and a data processing method are provided. The conference terminal includes: a mark obtaining module, configured to obtain a mark data signal; an image obtaining module, configured to obtain an image data signal; a superimposing module, configured to superimpose the mark data signal onto the image data signal to generate a hybrid image data signal; and an outputting module, configured to output the hybrid image data signal. The present invention improves the interactivity between videoconference users and the discussion efficiency.

Conference terminal, conference server, conference system and data processing method

A method, an apparatus, and a system for implementing audio mixing are provided. The method for implementing audio mixing includes: receiving an audio signal sent by each sending conference site, where the audio signals include an audio channel-based audio signal and an audio object-based audio signal; selecting an audio signal for each receiving conference site from the received audio signals; processing the selected audio signals according to types of receiving conference sites, where the receiving conference sites include an audio channel-based receiving conference site and an audio object-based receiving conference site; and sending the processed audio signal to each receiving conference site respectively according to the type of the receiving conference site. The present invention is applicable to a multipoint conference where audio mixing is performed on an audio channel-based audio signal and an audio object-based audio signal.

Method, apparatus, and system for implementing audio mixing

The embodiments of the present invention disclose a method, device, and system for video communication, and relate to the field of video conference technologies, so as to implement automatic switching of video images during a video conference. The method includes: obtaining video image signals and voice information of a first site; determining video image signals including a video object according to the video image signals and voice information of the first site; and sending the video image signals including the video object to a second site. The method, device, and system provided in the embodiments of the present invention implement automatic switching of video image signals during the video conference, thus avoiding switching errors caused by human factors and improving the efficiency of the conference.

Method, device, and system for video communication

A method for generating and playing audio signals and a system for processing audio signals are disclosed. The method for generating audio signals includes: generating distance information about an audio signal corresponding to a view point position, according to obtained auxiliary video and direction information about the audio signal, where the auxiliary video is a disparity map or a depth map; encoding the direction information and distance information about the audio signal, and sending the encoded information. The apparatus for generating audio signals includes an audio signal distance information obtaining module and an audio signal encoding module. With the present invention, the position information, including direction information and distance information, about the audio signal may be obtained accurately in combination with a three-dimensional video signal and a three-dimensional audio signal, without increasing the size of a microphone array, and the audio signal is sent and played.

Method and apparatus for generating and playing audio signals, and system for processing audio signals

A method, device and system to implement hiding the loss packet are provided. The provided method, device and system recover the lost frame according to the data before and after the lost frame and enhances the correlation of the recovered lost frame data and the data after the lost frame. A method and device for estimating pitch period are also provided which select a pitch period from the initial pitch period and the pitch periods corresponding to the frequencies which are one or more times higher than the frequencies corresponding to the initial pitch period as the final estimated pitch period, may improve frequency multiplication when estimating the pitch period; in addition, by tuning of the pitch period by matching the waves, the error of estimating pitch period may be reduced and the quality of the audio data may be improved.

Method, system, and device for performing packet loss concealment by superposing data

A method for concealing lost frame includes: using history signals before the lost frame that corresponds to a lost MDCT coefficient to generate a first synthesized signal when it is detected that the MDCT coefficient is lost; performing fast IMDCT for the first synthesized signal to obtain an IMDCT coefficient corresponding to a lost MDCT coefficient; and using the IMDCT coefficient corresponding to the lost MDCT coefficient and an IMDCT coefficient adjacent to the IMDCT coefficient corresponding to the lost MDCT coefficient to perform TDAC and obtain signals corresponding to the lost frame. An apparatus for concealing lost frame is also disclosed herein. The method and the apparatus for concealing lost frames in the embodiments of the present invention make full use of the received partial signals to recover high-quality voice signals and improve the QoS.

Method and apparatus for concealing lost frame

A device for lost frame concealment comprises: a lost frame detector for detecting whether a voice frame is lost, a decoding module for decoding the current voice frame, a low band delay module for delaying the low band signal, a low band signal recovering module for recovering the lost low band signal, a high band lost frame concealment module for processing the lost frame concealment for the high band signal, and a QMF synthesis filter for synthetically filtering the low band signal and the high band signal. The invention makes full use of the delay of the coding/decoding device itself, enhances the effect of lost frame concealment for the low band signal and the high band signal, and introduces no nearby delay during the process of lost frame concealment.

Method of lost frame concealment and device

The present invention discloses a method for obtaining an attenuation factor. The method is adapted to process the synthesized signal in packet loss concealment, and includes: obtaining a change trend of a signal; obtaining an attenuation factor, according to the change trend of the signal. The present invention also discloses an apparatus for obtaining an attenuation factor. A self-adaptive attenuation factor is adjusted dynamically by using the latest change trend of a history signal by using the present invention. The smooth transition from the history data to the data last received is realized so that the attenuation speed is kept consistent between the compensated signal and the original signal as much as possible for adapting to the characteristic of various human voices.

Method and apparatus for obtaining an attenuation factor

The present invention discloses a method for performing a frame erasure concealment to a higher-band signal, including: calculating a periodic intensity of a higher-band signal with respect to a lower-band signal; judging whether the periodic intensity of the higher-band signal is higher than or equal to a preconfigured threshold; if the periodic intensity of the higher-band signal is higher than or equal to the preconfigured threshold, using a pitch period repetition method to perform the frame erasure concealment to the higher-band signal of a current lost frame; and if the periodic intensity of the higher-band signal is lower than the preconfigured threshold, using a previous frame data repetition method to perform the frame erasure concealment to the higher-band signal of the current lost frame. The present invention further discloses a device for performing a frame erasure concealment to a higher-band signal and a speech decoder. The problem that the quality of the voice signal is lowered is avoided.

Method and device for performing frame erasure concealment to higher-band signal

The present invention discloses a method for obtaining an attenuation factor. The method is adapted to process the synthesized signal in packet loss concealment, and includes: obtaining a change trend of a pitch of a signal; obtaining an attenuation factor, according to the change trend of the pitch of the signal. The present invention also discloses an apparatus for obtaining an attenuation factor. A self-adaptive attenuation factor is adjusted dynamically by using the latest change trend of a history signal by using the present invention. The smooth transition from the history data to the data last received is realized so that the attenuation speed is kept consistent between the compensated signal and the original signal as much as possible for adapting to the characteristic of various human voices.

The present invention discloses a signal processing method adapted to process a synthesized signal in packet loss concealment. The method includes the following steps: receiving a good frame following a lost frame, obtaining an energy ratio of energy of a signal in the signal of the good frame signal to energy of a synthesized signal corresponding to the same time of the good frame; and adjusting the synthesized signal in accordance with the energy ratio. The present invention also discloses a signal processing apparatus and a voice decoder. Through using the method provided by the present invention, the synthesized signal is adjusted in accordance with the energy ratio of the energy of the first good frame following the lost frame to the energy of the synthesized signal to ensure that there be not a waveform sudden change or an energy sudden change at the place where the lost frame and the first good frame following the lost frame are jointed in the synthesized signal, to realize the waveform's smooth transition and to avoid music noises.

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Shenzhen, China

Dongqi Wang