Abstract: 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.

Abstract: 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.

Abstract: The present invention discloses an image controlling method, device, and system for a composed-image video conference, where the method includes receiving audio data of sites; obtaining, according to audio data of each site of the sites and in real time, a voice feature value that is within a first specified period and of a corresponding site, where the voice feature value is used to represent an activation state of the site; selecting a specified site from the multiple sites according to an activation state of each site; and filling a picture of the specified site into a sub-image of a composed image, to update the composed image in real time. This remarkably improves an effect of a conference, and improves experience of participants. In addition, a quantity and locations of sub-images in the composed image may be further adjusted dynamically, which also effectively improves the effect of the conference.

Abstract: 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.

Abstract: 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.

Abstract: Embodiments of the present invention provide a method, device, and system for classifying voice conference minutes. The method is: performing voice source locating according to audio data of the conference site so as to acquire a location of a voice source corresponding to the audio data, writing the location of the voice source into additional field information of the audio data, writing a voice activation flag into the additional field information, packaging the audio data as an audio code stream, and sending the audio code stream and the additional field information of the audio code stream to a recording server, so that the recording server classifies the audio data according to the additional field information and writes a participant identity that corresponds to the location of the voice source corresponding to the audio data into the additional field information of the audio code stream.

Abstract: The present invention discloses an echo cancellation method. The method includes: dividing an audio signal into a high-band audio signal and a low-band audio signal; performing adaptive filtering on the low-band audio signal, and performing synthesis filtering on a signal obtained after the low-band audio signal undergoes the adaptive filtering and on the high-band audio signal to generate a preliminary echo cancellation signal; performing envelope predication echo suppression on a high-band signal in the preliminary echo cancellation signal, and calculating and outputting a residual echo suppression coefficient; performing echo suppression on a low-band signal in the preliminary echo cancellation signal, and outputting a processing result; and multiplying the output result by the residual echo suppression coefficient, and outputting a signal of which echoes are canceled.

Abstract: 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.

Abstract: 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.

Abstract: Embodiments of the present invention provide a method, device, and system for classifying voice conference minutes. The method is: performing voice source locating according to audio data of the conference site so as to acquire a location of a voice source corresponding to the audio data, writing the location of the voice source into additional field information of the audio data, writing a voice activation flag into the additional field information, packaging the audio data as an audio code stream, and sending the audio code stream and the additional field information of the audio code stream to a recording server, so that the recording server classifies the audio data according to the additional field information and writes a participant identity that corresponds to the location of the voice source corresponding to the audio data into the additional field information of the audio code stream.

Abstract: 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.

Abstract: 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.

Abstract: The present invention discloses an echo cancellation method. The method includes: dividing an audio signal into a high-band audio signal and a low-band audio signal; performing adaptive filtering on the low-band audio signal, and performing synthesis filtering on a signal obtained after the low-band audio signal undergoes the adaptive filtering and on the high-band audio signal to generate a preliminary echo cancellation signal; performing envelope predication echo suppression on a high-band signal in the preliminary echo cancellation signal, and calculating and outputting a residual echo suppression coefficient; performing echo suppression on a low-band signal in the preliminary echo cancellation signal, and outputting a processing result; and multiplying the output result by the residual echo suppression coefficient, and outputting a signal of which echoes are canceled.

Abstract: 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.

Abstract: Embodiments of the present invention provide a multiple description audio coding and decoding method, apparatus, and system. The audio coding method includes: dividing residual signals indicating current audio signal information into multiple frequency band parts having different frequencies; respectively coding the multiple frequency band parts by using multiple description coding (MDC) methods with different speech quality; and combining each of description signal parts that are generated after coding is performed by using different MDC methods to form multiple description bit streams of the residual signals. According to the present invention, multiple description coding and decoding methods with different speech quality are used for different frequency bands, which reduces the bit rate of multiple description coding and decoding, improves the effect of multiple description coding and decoding, and hence enhances the quality of audio transmission.

Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: A method, apparatus and system for multiple-description coding (MDC) and multiple-description decoding (MDD) are provided in the present invention. The MDC method comprises the following steps: receiving signals, processing the signal according to the first multiple-description method to generate at least two first description signals, processing at least one of the first description signals through the second multiple-description method to generate at least two second description signals, and coding the second description signal to generate the multiple-description bit stream.

Abstract: 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.